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authorLinus Torvalds <torvalds@linux-foundation.org>2009-09-11 09:19:35 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2009-09-11 09:19:35 -0700
commita9c86d42599519f3d83b5f46bdab25046fe47b84 (patch)
tree9b269e3162e5cc0c1a8dfc3349303c902718a9a9
parenta12e4d304ce701844c639541d90df86e165d03f9 (diff)
parent1110afbe728838ac7ce973c37af9e11385dbaef9 (diff)
downloadop-kernel-dev-a9c86d42599519f3d83b5f46bdab25046fe47b84.zip
op-kernel-dev-a9c86d42599519f3d83b5f46bdab25046fe47b84.tar.gz
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (377 commits) ASoC: au1x: PSC-AC97 bugfixes ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128 ALSA: dummy - Add debug proc file ALSA: Add const prefix to proc helper functions ALSA: Re-export snd_pcm_format_name() function ALSA: hda - Use auto model for HP laptops with ALC268 codec ALSA: cs46xx - Fix minimum period size ASoC: Fix WM835x Out4 capture enumeration ALSA: Remove unneeded ifdef from sound/core.h ALSA: Remove struct snd_monitor_file from public sound/core.h ASoC: Remove unuused hw_read_t sound: oxygen: work around MCE when changing volume ALSA: dummy - Fake buffer allocations ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital ASoC: fix pxa2xx-ac97.c breakage ALSA: dummy - Fix the timer calculation in systimer mode ALSA: dummy - Add more description ALSA: dummy - Better jiffies handling ALSA: dummy - Support high-res timer mode ALSA: Release v1.0.21 ...
-rw-r--r--Documentation/feature-removal-schedule.txt24
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt30
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt33
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt64
-rw-r--r--arch/arm/mach-omap2/mcbsp.c5
-rw-r--r--arch/arm/mach-pxa/include/mach/audio.h3
-rw-r--r--arch/arm/plat-omap/dma.c10
-rw-r--r--arch/arm/plat-omap/include/mach/mcbsp.h51
-rw-r--r--arch/arm/plat-omap/mcbsp.c401
-rw-r--r--arch/arm/plat-s3c/include/plat/audio-simtec.h37
-rw-r--r--arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h5
-rw-r--r--fs/char_dev.c39
-rw-r--r--include/linux/fs.h19
-rw-r--r--include/linux/tty.h4
-rw-r--r--include/sound/ac97_codec.h9
-rw-r--r--include/sound/asound.h2
-rw-r--r--include/sound/core.h51
-rw-r--r--include/sound/info.h4
-rw-r--r--include/sound/memalloc.h6
-rw-r--r--include/sound/pcm.h25
-rw-r--r--include/sound/sh_fsi.h83
-rw-r--r--include/sound/soc-dai.h40
-rw-r--r--include/sound/soc-dapm.h10
-rw-r--r--include/sound/soc.h49
-rw-r--r--include/sound/tlv.h14
-rw-r--r--include/sound/uda1380.h22
-rw-r--r--include/sound/version.h2
-rw-r--r--include/sound/wm8993.h44
-rw-r--r--include/sound/ymfpci.h1
-rw-r--r--sound/Kconfig28
-rw-r--r--sound/arm/pxa2xx-ac97.c10
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c3
-rw-r--r--sound/core/Kconfig4
-rw-r--r--sound/core/Makefile2
-rw-r--r--sound/core/control.c34
-rw-r--r--sound/core/info.c8
-rw-r--r--sound/core/init.c8
-rw-r--r--sound/core/memalloc.c4
-rw-r--r--sound/core/misc.c75
-rw-r--r--sound/core/oss/mixer_oss.c3
-rw-r--r--sound/core/oss/pcm_oss.c12
-rw-r--r--sound/core/pcm.c26
-rw-r--r--sound/core/pcm_lib.c12
-rw-r--r--sound/core/pcm_memory.c2
-rw-r--r--sound/core/pcm_native.c64
-rw-r--r--sound/core/rawmidi.c2
-rw-r--r--sound/core/seq/oss/seq_oss_midi.c14
-rw-r--r--sound/core/seq/seq_midi.c7
-rw-r--r--sound/core/vmaster.c8
-rw-r--r--sound/drivers/dummy.c700
-rw-r--r--sound/isa/cmi8330.c86
-rw-r--r--sound/oss/midibuf.c7
-rw-r--r--sound/oss/vwsnd.c6
-rw-r--r--sound/pci/Kconfig4
-rw-r--r--sound/pci/ali5451/ali5451.c65
-rw-r--r--sound/pci/azt3328.c1116
-rw-r--r--sound/pci/azt3328.h103
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.h2
-rw-r--r--sound/pci/ctxfi/ct20k2reg.h9
-rw-r--r--sound/pci/ctxfi/ctamixer.c20
-rw-r--r--sound/pci/ctxfi/ctatc.c77
-rw-r--r--sound/pci/ctxfi/ctdaio.c30
-rw-r--r--sound/pci/ctxfi/cthw20k1.c22
-rw-r--r--sound/pci/ctxfi/cthw20k2.c73
-rw-r--r--sound/pci/ctxfi/ctmixer.c8
-rw-r--r--sound/pci/ctxfi/ctpcm.c6
-rw-r--r--sound/pci/ctxfi/ctresource.c4
-rw-r--r--sound/pci/ctxfi/ctsrc.c10
-rw-r--r--sound/pci/ctxfi/ctvmem.c6
-rw-r--r--sound/pci/hda/Kconfig27
-rw-r--r--sound/pci/hda/Makefile4
-rw-r--r--sound/pci/hda/hda_beep.c4
-rw-r--r--sound/pci/hda/hda_codec.c68
-rw-r--r--sound/pci/hda/hda_codec.h10
-rw-r--r--sound/pci/hda/hda_generic.c18
-rw-r--r--sound/pci/hda/hda_hwdep.c236
-rw-r--r--sound/pci/hda/hda_intel.c74
-rw-r--r--sound/pci/hda/hda_local.h14
-rw-r--r--sound/pci/hda/hda_proc.c7
-rw-r--r--sound/pci/hda/patch_analog.c131
-rw-r--r--sound/pci/hda/patch_atihdmi.c3
-rw-r--r--sound/pci/hda/patch_ca0110.c3
-rw-r--r--sound/pci/hda/patch_cirrus.c1194
-rw-r--r--sound/pci/hda/patch_cmedia.c3
-rw-r--r--sound/pci/hda/patch_conexant.c479
-rw-r--r--sound/pci/hda/patch_intelhdmi.c104
-rw-r--r--sound/pci/hda/patch_nvhdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c4144
-rw-r--r--sound/pci/hda/patch_sigmatel.c1206
-rw-r--r--sound/pci/hda/patch_via.c3
-rw-r--r--sound/pci/ice1712/ice1712.h9
-rw-r--r--sound/pci/ice1712/ice1724.c112
-rw-r--r--sound/pci/ice1712/prodigy_hifi.c46
-rw-r--r--sound/pci/oxygen/oxygen_io.c11
-rw-r--r--sound/pci/rme9652/hdsp.c39
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c20
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile3
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c138
-rw-r--r--sound/soc/au1x/psc-ac97.c129
-rw-r--r--sound/soc/au1x/psc.h1
-rw-r--r--sound/soc/blackfin/Kconfig31
-rw-r--r--sound/soc/blackfin/Makefile8
-rw-r--r--sound/soc/blackfin/bf5xx-ac97.c10
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c128
-rw-r--r--sound/soc/blackfin/bf5xx-ad1938.c142
-rw-r--r--sound/soc/blackfin/bf5xx-ad73311.c16
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c8
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c16
-rw-r--r--sound/soc/blackfin/bf5xx-tdm-pcm.c330
-rw-r--r--sound/soc/blackfin/bf5xx-tdm-pcm.h21
-rw-r--r--sound/soc/blackfin/bf5xx-tdm.c343
-rw-r--r--sound/soc/blackfin/bf5xx-tdm.h14
-rw-r--r--sound/soc/codecs/Kconfig44
-rw-r--r--sound/soc/codecs/Makefile26
-rw-r--r--sound/soc/codecs/ad1836.c446
-rw-r--r--sound/soc/codecs/ad1836.h64
-rw-r--r--sound/soc/codecs/ad1938.c682
-rw-r--r--sound/soc/codecs/ad1938.h100
-rw-r--r--sound/soc/codecs/ak4535.c16
-rw-r--r--sound/soc/codecs/ak4642.c502
-rw-r--r--sound/soc/codecs/ak4642.h20
-rw-r--r--sound/soc/codecs/cs4270.c27
-rw-r--r--sound/soc/codecs/cx20442.c501
-rw-r--r--sound/soc/codecs/cx20442.h20
-rw-r--r--sound/soc/codecs/max9877.c308
-rw-r--r--sound/soc/codecs/max9877.h37
-rw-r--r--sound/soc/codecs/spdif_transciever.c3
-rw-r--r--sound/soc/codecs/stac9766.c4
-rw-r--r--sound/soc/codecs/tlv320aic3x.c233
-rw-r--r--sound/soc/codecs/tlv320aic3x.h2
-rw-r--r--sound/soc/codecs/twl4030.c260
-rw-r--r--sound/soc/codecs/twl4030.h2
-rw-r--r--sound/soc/codecs/uda134x.c2
-rw-r--r--sound/soc/codecs/uda1380.c313
-rw-r--r--sound/soc/codecs/uda1380.h8
-rw-r--r--sound/soc/codecs/wm8350.c51
-rw-r--r--sound/soc/codecs/wm8400.c26
-rw-r--r--sound/soc/codecs/wm8510.c175
-rw-r--r--sound/soc/codecs/wm8523.c699
-rw-r--r--sound/soc/codecs/wm8523.h160
-rw-r--r--sound/soc/codecs/wm8580.c211
-rw-r--r--sound/soc/codecs/wm8728.c111
-rw-r--r--sound/soc/codecs/wm8731.c218
-rw-r--r--sound/soc/codecs/wm8750.c154
-rw-r--r--sound/soc/codecs/wm8753.c35
-rw-r--r--sound/soc/codecs/wm8776.c744
-rw-r--r--sound/soc/codecs/wm8776.h51
-rw-r--r--sound/soc/codecs/wm8900.c345
-rw-r--r--sound/soc/codecs/wm8903.c267
-rw-r--r--sound/soc/codecs/wm8940.c160
-rw-r--r--sound/soc/codecs/wm8960.c233
-rw-r--r--sound/soc/codecs/wm8961.c1265
-rw-r--r--sound/soc/codecs/wm8961.h866
-rw-r--r--sound/soc/codecs/wm8971.c127
-rw-r--r--sound/soc/codecs/wm8974.c808
-rw-r--r--sound/soc/codecs/wm8974.h99
-rw-r--r--sound/soc/codecs/wm8988.c180
-rw-r--r--sound/soc/codecs/wm8990.c194
-rw-r--r--sound/soc/codecs/wm8993.c1675
-rw-r--r--sound/soc/codecs/wm8993.h2132
-rw-r--r--sound/soc/codecs/wm9081.c317
-rw-r--r--sound/soc/codecs/wm9705.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c743
-rw-r--r--sound/soc/codecs/wm_hubs.h24
-rw-r--r--sound/soc/davinci/Kconfig33
-rw-r--r--sound/soc/davinci/Makefile5
-rw-r--r--sound/soc/davinci/davinci-evm.c140
-rw-r--r--sound/soc/davinci/davinci-i2s.c340
-rw-r--r--sound/soc/davinci/davinci-mcasp.c973
-rw-r--r--sound/soc/davinci/davinci-mcasp.h60
-rw-r--r--sound/soc/davinci/davinci-pcm.c10
-rw-r--r--sound/soc/davinci/davinci-pcm.h19
-rw-r--r--sound/soc/fsl/mpc5200_dma.c17
-rw-r--r--sound/soc/fsl/mpc5200_psc_ac97.c3
-rw-r--r--sound/soc/imx/Kconfig21
-rw-r--r--sound/soc/imx/Makefile10
-rw-r--r--sound/soc/imx/mx1_mx2-pcm.c488
-rw-r--r--sound/soc/imx/mx1_mx2-pcm.h26
-rw-r--r--sound/soc/imx/mx27vis_wm8974.c317
-rw-r--r--sound/soc/imx/mxc-ssi.c868
-rw-r--r--sound/soc/imx/mxc-ssi.h238
-rw-r--r--sound/soc/omap/Kconfig15
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/ams-delta.c646
-rw-r--r--sound/soc/omap/n810.c12
-rw-r--r--sound/soc/omap/omap-mcbsp.c123
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/omap/omap-pcm.c53
-rw-r--r--sound/soc/omap/omap-pcm.h2
-rw-r--r--sound/soc/omap/sdp3430.c18
-rw-r--r--sound/soc/omap/zoom2.c314
-rw-r--r--sound/soc/pxa/magician.c56
-rw-r--r--sound/soc/pxa/palm27x.c204
-rw-r--r--sound/soc/pxa/pxa-ssp.c77
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c12
-rw-r--r--sound/soc/s3c24xx/Kconfig35
-rw-r--r--sound/soc/s3c24xx/Makefile9
-rw-r--r--sound/soc/s3c24xx/neo1973_gta02_wm8753.c498
-rw-r--r--sound/soc/s3c24xx/s3c-i2s-v2.c17
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c20
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c5
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c2
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.c394
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.h22
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_hermes.c153
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c137
-rw-r--r--sound/soc/s6000/s6105-ipcam.c12
-rw-r--r--sound/soc/sh/Kconfig15
-rw-r--r--sound/soc/sh/Makefile4
-rw-r--r--sound/soc/sh/fsi-ak4642.c107
-rw-r--r--sound/soc/sh/fsi.c1004
-rw-r--r--sound/soc/soc-cache.c218
-rw-r--r--sound/soc/soc-core.c148
-rw-r--r--sound/soc/soc-dapm.c498
-rw-r--r--sound/soc/soc-jack.c24
-rw-r--r--sound/soc/txx9/txx9aclc.c10
-rw-r--r--sound/sound_core.c100
-rw-r--r--sound/usb/usbaudio.c6
-rw-r--r--sound/usb/usbmidi.c290
-rw-r--r--sound/usb/usbmixer.c73
221 files changed, 30709 insertions, 6969 deletions
diff --git a/Documentation/feature-removal-schedule.txt b/Documentation/feature-removal-schedule.txt
index 09e031c..f0690bb 100644
--- a/Documentation/feature-removal-schedule.txt
+++ b/Documentation/feature-removal-schedule.txt
@@ -468,3 +468,27 @@ Why: cpu_policy_rwsem has a new cleaner definition making it local to
cpufreq core and contained inside cpufreq.c. Other dependent
drivers should not use it in order to safely avoid lockdep issues.
Who: Venkatesh Pallipadi <venkatesh.pallipadi@intel.com>
+
+----------------------------
+
+What: sound-slot/service-* module aliases and related clutters in
+ sound/sound_core.c
+When: August 2010
+Why: OSS sound_core grabs all legacy minors (0-255) of SOUND_MAJOR
+ (14) and requests modules using custom sound-slot/service-*
+ module aliases. The only benefit of doing this is allowing
+ use of custom module aliases which might as well be considered
+ a bug at this point. This preemptive claiming prevents
+ alternative OSS implementations.
+
+ Till the feature is removed, the kernel will be requesting
+ both sound-slot/service-* and the standard char-major-* module
+ aliases and allow turning off the pre-claiming selectively via
+ CONFIG_SOUND_OSS_CORE_PRECLAIM and soundcore.preclaim_oss
+ kernel parameter.
+
+ After the transition phase is complete, both the custom module
+ aliases and switches to disable it will go away. This removal
+ will also allow making ALSA OSS emulation independent of
+ sound_core. The dependency will be broken then too.
+Who: Tejun Heo <tj@kernel.org>
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 4252697..1c8eb45 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -60,6 +60,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
slots - Reserve the slot index for the given driver.
This option takes multiple strings.
See "Module Autoloading Support" section for details.
+ debug - Specifies the debug message level
+ (0 = disable debug prints, 1 = normal debug messages,
+ 2 = verbose debug messages)
+ This option appears only when CONFIG_SND_DEBUG=y.
+ This option can be dynamically changed via sysfs
+ /sys/modules/snd/parameters/debug file.
Module snd-pcm-oss
------------------
@@ -513,6 +519,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
or input, but you may use this module for any application which
requires a sound card (like RealPlayer).
+ pcm_devs - Number of PCM devices assigned to each card
+ (default = 1, up to 4)
+ pcm_substreams - Number of PCM substreams assigned to each PCM
+ (default = 8, up to 16)
+ hrtimer - Use hrtimer (=1, default) or system timer (=0)
+ fake_buffer - Fake buffer allocations (default = 1)
+
+ When multiple PCM devices are created, snd-dummy gives different
+ behavior to each PCM device:
+ 0 = interleaved with mmap support
+ 1 = non-interleaved with mmap support
+ 2 = interleaved without mmap
+ 3 = non-interleaved without mmap
+
+ As default, snd-dummy drivers doesn't allocate the real buffers
+ but either ignores read/write or mmap a single dummy page to all
+ buffer pages, in order to save the resouces. If your apps need
+ the read/ written buffer data to be consistent, pass fake_buffer=0
+ option.
+
The power-management is supported.
Module snd-echo3g
@@ -768,6 +794,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
bdl_pos_adj - Specifies the DMA IRQ timing delay in samples.
Passing -1 will make the driver to choose the appropriate
value based on the controller chip.
+ patch - Specifies the early "patch" files to modify the HD-audio
+ setup before initializing the codecs. This option is
+ available only when CONFIG_SND_HDA_PATCH_LOADER=y is set.
+ See HD-Audio.txt for details.
[Single (global) options]
single_cmd - Use single immediate commands to communicate with
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 939a3dd..97eebd6 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -114,8 +114,8 @@ ALC662/663/272
samsung-nc10 Samsung NC10 mini notebook
auto auto-config reading BIOS (default)
-ALC882/885
-==========
+ALC882/883/885/888/889
+======================
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
@@ -127,12 +127,8 @@ ALC882/885
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
- auto auto-config reading BIOS (default)
-
-ALC883/888
-==========
- 3stack-dig 3-jack with SPDIF I/O
- 6stack-dig 6-jack digital with SPDIF I/O
+ 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883)
+ alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883)
3stack-6ch 3-jack 6-channel
3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
6stack-dig-demo 6-jack digital for Intel demo board
@@ -140,6 +136,7 @@ ALC883/888
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
acer-aspire-6530g Acer Aspire 6530G
+ acer-aspire-7730g Acer Aspire 7730G
acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
medion-md2 Medion MD2
@@ -155,10 +152,13 @@ ALC883/888
3stack-hp HP machines with 3stack (Lucknow, Samba boards)
6stack-dell Dell machines with 6stack (Inspiron 530)
mitac Mitac 8252D
+ clevo-m540r Clevo M540R (6ch + digital)
clevo-m720 Clevo M720 laptop series
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
+ intel-alc889a Intel IbexPeak with ALC889A
+ intel-x58 Intel DX58 with ALC889
asus-p5q ASUS P5Q-EM boards
mb31 MacBook 3,1
sony-vaio-tt Sony VAIO TT
@@ -229,7 +229,7 @@ AD1984
======
basic default configuration
thinkpad Lenovo Thinkpad T61/X61
- dell Dell T3400
+ dell_desktop Dell T3400
AD1986A
=======
@@ -258,6 +258,7 @@ Conexant 5045
laptop-micsense Laptop with Mic sense (old model fujitsu)
laptop-hpmicsense Laptop with HP and Mic senses
benq Benq R55E
+ laptop-hp530 HP 530 laptop
test for testing/debugging purpose, almost all controls
can be adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
@@ -278,9 +279,16 @@ Conexant 5051
hp-dv6736 HP dv6736
lenovo-x200 Lenovo X200 laptop
+Conexant 5066
+=============
+ laptop Basic Laptop config (default)
+ dell-laptop Dell laptops
+ olpc-xo-1_5 OLPC XO 1.5
+
STAC9200
========
ref Reference board
+ oqo OQO Model 2
dell-d21 Dell (unknown)
dell-d22 Dell (unknown)
dell-d23 Dell (unknown)
@@ -368,10 +376,12 @@ STAC92HD73*
===========
ref Reference board
no-jd BIOS setup but without jack-detection
+ intel Intel DG45* mobos
dell-m6-amic Dell desktops/laptops with analog mics
dell-m6-dmic Dell desktops/laptops with digital mics
dell-m6 Dell desktops/laptops with both type of mics
dell-eq Dell desktops/laptops
+ alienware Alienware M17x
auto BIOS setup (default)
STAC92HD83*
@@ -385,3 +395,8 @@ STAC9872
========
vaio VAIO laptop without SPDIF
auto BIOS setup (default)
+
+Cirrus Logic CS4206/4207
+========================
+ mbp55 MacBook Pro 5,5
+ auto BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 71ac995..7b8a5f9 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -139,6 +139,10 @@ The driver checks PCI SSID and looks through the static configuration
table until any matching entry is found. If you have a new machine,
you may see a message like below:
------------------------------------------------------------------------
+ hda_codec: ALC880: BIOS auto-probing.
+------------------------------------------------------------------------
+Meanwhile, in the earlier versions, you would see a message like:
+------------------------------------------------------------------------
hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
------------------------------------------------------------------------
Even if you see such a message, DON'T PANIC. Take a deep breath and
@@ -403,6 +407,66 @@ re-configure based on that state, run like below:
------------------------------------------------------------------------
+Early Patching
+~~~~~~~~~~~~~~
+When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
+firmware file for modifying the HD-audio setup before initializing the
+codec. This can work basically like the reconfiguration via sysfs in
+the above, but it does it before the first codec configuration.
+
+A patch file is a plain text file which looks like below:
+
+------------------------------------------------------------------------
+ [codec]
+ 0x12345678 0xabcd1234 2
+
+ [model]
+ auto
+
+ [pincfg]
+ 0x12 0x411111f0
+
+ [verb]
+ 0x20 0x500 0x03
+ 0x20 0x400 0xff
+
+ [hint]
+ hp_detect = yes
+------------------------------------------------------------------------
+
+The file needs to have a line `[codec]`. The next line should contain
+three numbers indicating the codec vendor-id (0x12345678 in the
+example), the codec subsystem-id (0xabcd1234) and the address (2) of
+the codec. The rest patch entries are applied to this specified codec
+until another codec entry is given.
+
+The `[model]` line allows to change the model name of the each codec.
+In the example above, it will be changed to model=auto.
+Note that this overrides the module option.
+
+After the `[pincfg]` line, the contents are parsed as the initial
+default pin-configurations just like `user_pin_configs` sysfs above.
+The values can be shown in user_pin_configs sysfs file, too.
+
+Similarly, the lines after `[verb]` are parsed as `init_verbs`
+sysfs entries, and the lines after `[hint]` are parsed as `hints`
+sysfs entries, respectively.
+
+The hd-audio driver reads the file via request_firmware(). Thus,
+a patch file has to be located on the appropriate firmware path,
+typically, /lib/firmware. For example, when you pass the option
+`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
+present.
+
+The patch module option is specific to each card instance, and you
+need to give one file name for each instance, separated by commas.
+For example, if you have two cards, one for an on-board analog and one
+for an HDMI video board, you may pass patch option like below:
+------------------------------------------------------------------------
+ options snd-hda-intel patch=on-board-patch,hdmi-patch
+------------------------------------------------------------------------
+
+
Power-Saving
~~~~~~~~~~~~
The power-saving is a kind of auto-suspend of the device. When the
diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c
index 99b6e15..0447d26 100644
--- a/arch/arm/mach-omap2/mcbsp.c
+++ b/arch/arm/mach-omap2/mcbsp.c
@@ -128,6 +128,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = {
.rx_irq = INT_24XX_MCBSP1_IRQ_RX,
.tx_irq = INT_24XX_MCBSP1_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP2_BASE,
@@ -136,6 +137,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = {
.rx_irq = INT_24XX_MCBSP2_IRQ_RX,
.tx_irq = INT_24XX_MCBSP2_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x3FF,
},
{
.phys_base = OMAP34XX_MCBSP3_BASE,
@@ -144,6 +146,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = {
.rx_irq = INT_24XX_MCBSP3_IRQ_RX,
.tx_irq = INT_24XX_MCBSP3_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP4_BASE,
@@ -152,6 +155,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = {
.rx_irq = INT_24XX_MCBSP4_IRQ_RX,
.tx_irq = INT_24XX_MCBSP4_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
{
.phys_base = OMAP34XX_MCBSP5_BASE,
@@ -160,6 +164,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = {
.rx_irq = INT_24XX_MCBSP5_IRQ_RX,
.tx_irq = INT_24XX_MCBSP5_IRQ_TX,
.ops = &omap2_mcbsp_ops,
+ .buffer_size = 0x6F,
},
};
#define OMAP34XX_MCBSP_PDATA_SZ ARRAY_SIZE(omap34xx_mcbsp_pdata)
diff --git a/arch/arm/mach-pxa/include/mach/audio.h b/arch/arm/mach-pxa/include/mach/audio.h
index 16eb025..a3449e35a 100644
--- a/arch/arm/mach-pxa/include/mach/audio.h
+++ b/arch/arm/mach-pxa/include/mach/audio.h
@@ -3,10 +3,12 @@
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
/*
* @reset_gpio: AC97 reset gpio (normally gpio113 or gpio95)
* a -1 value means no gpio will be used for reset
+ * @codec_pdata: AC97 codec platform_data
* reset_gpio should only be specified for pxa27x CPUs where a silicon
* bug prevents correct operation of the reset line. If not specified,
@@ -20,6 +22,7 @@ typedef struct {
void (*resume)(void *);
void *priv;
int reset_gpio;
+ void *codec_pdata[AC97_BUS_MAX_DEVICES];
} pxa2xx_audio_ops_t;
extern void pxa_set_ac97_info(pxa2xx_audio_ops_t *ops);
diff --git a/arch/arm/plat-omap/dma.c b/arch/arm/plat-omap/dma.c
index e3ac94f..9b00f4c 100644
--- a/arch/arm/plat-omap/dma.c
+++ b/arch/arm/plat-omap/dma.c
@@ -1127,6 +1127,11 @@ int omap_dma_running(void)
void omap_dma_link_lch(int lch_head, int lch_queue)
{
if (omap_dma_in_1510_mode()) {
+ if (lch_head == lch_queue) {
+ dma_write(dma_read(CCR(lch_head)) | (3 << 8),
+ CCR(lch_head));
+ return;
+ }
printk(KERN_ERR "DMA linking is not supported in 1510 mode\n");
BUG();
return;
@@ -1149,6 +1154,11 @@ EXPORT_SYMBOL(omap_dma_link_lch);
void omap_dma_unlink_lch(int lch_head, int lch_queue)
{
if (omap_dma_in_1510_mode()) {
+ if (lch_head == lch_queue) {
+ dma_write(dma_read(CCR(lch_head)) & ~(3 << 8),
+ CCR(lch_head));
+ return;
+ }
printk(KERN_ERR "DMA linking is not supported in 1510 mode\n");
BUG();
return;
diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h
index bb154ea..63a3f25 100644
--- a/arch/arm/plat-omap/include/mach/mcbsp.h
+++ b/arch/arm/plat-omap/include/mach/mcbsp.h
@@ -134,6 +134,11 @@
#define OMAP_MCBSP_REG_XCERG 0x74
#define OMAP_MCBSP_REG_XCERH 0x78
#define OMAP_MCBSP_REG_SYSCON 0x8C
+#define OMAP_MCBSP_REG_THRSH2 0x90
+#define OMAP_MCBSP_REG_THRSH1 0x94
+#define OMAP_MCBSP_REG_IRQST 0xA0
+#define OMAP_MCBSP_REG_IRQEN 0xA4
+#define OMAP_MCBSP_REG_WAKEUPEN 0xA8
#define OMAP_MCBSP_REG_XCCR 0xAC
#define OMAP_MCBSP_REG_RCCR 0xB0
@@ -249,8 +254,27 @@
#define RDISABLE 0x0001
/********************** McBSP SYSCONFIG bit definitions ********************/
+#define CLOCKACTIVITY(value) ((value)<<8)
+#define SIDLEMODE(value) ((value)<<3)
+#define ENAWAKEUP 0x0004
#define SOFTRST 0x0002
+/********************** McBSP DMA operating modes **************************/
+#define MCBSP_DMA_MODE_ELEMENT 0
+#define MCBSP_DMA_MODE_THRESHOLD 1
+#define MCBSP_DMA_MODE_FRAME 2
+
+/********************** McBSP WAKEUPEN bit definitions *********************/
+#define XEMPTYEOFEN 0x4000
+#define XRDYEN 0x0400
+#define XEOFEN 0x0200
+#define XFSXEN 0x0100
+#define XSYNCERREN 0x0080
+#define RRDYEN 0x0008
+#define REOFEN 0x0004
+#define RFSREN 0x0002
+#define RSYNCERREN 0x0001
+
/* we don't do multichannel for now */
struct omap_mcbsp_reg_cfg {
u16 spcr2;
@@ -344,6 +368,9 @@ struct omap_mcbsp_platform_data {
u8 dma_rx_sync, dma_tx_sync;
u16 rx_irq, tx_irq;
struct omap_mcbsp_ops *ops;
+#ifdef CONFIG_ARCH_OMAP34XX
+ u16 buffer_size;
+#endif
};
struct omap_mcbsp {
@@ -377,6 +404,11 @@ struct omap_mcbsp {
struct omap_mcbsp_platform_data *pdata;
struct clk *iclk;
struct clk *fclk;
+#ifdef CONFIG_ARCH_OMAP34XX
+ int dma_op_mode;
+ u16 max_tx_thres;
+ u16 max_rx_thres;
+#endif
};
extern struct omap_mcbsp **mcbsp_ptr;
extern int omap_mcbsp_count;
@@ -385,10 +417,25 @@ int omap_mcbsp_init(void);
void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config,
int size);
void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config);
+#ifdef CONFIG_ARCH_OMAP34XX
+void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold);
+void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold);
+u16 omap_mcbsp_get_max_tx_threshold(unsigned int id);
+u16 omap_mcbsp_get_max_rx_threshold(unsigned int id);
+int omap_mcbsp_get_dma_op_mode(unsigned int id);
+#else
+static inline void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold)
+{ }
+static inline void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold)
+{ }
+static inline u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) { return 0; }
+static inline u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) { return 0; }
+static inline int omap_mcbsp_get_dma_op_mode(unsigned int id) { return 0; }
+#endif
int omap_mcbsp_request(unsigned int id);
void omap_mcbsp_free(unsigned int id);
-void omap_mcbsp_start(unsigned int id);
-void omap_mcbsp_stop(unsigned int id);
+void omap_mcbsp_start(unsigned int id, int tx, int rx);
+void omap_mcbsp_stop(unsigned int id, int tx, int rx);
void omap_mcbsp_xmit_word(unsigned int id, u32 word);
u32 omap_mcbsp_recv_word(unsigned int id);
diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c
index efa0e01..8dc7927 100644
--- a/arch/arm/plat-omap/mcbsp.c
+++ b/arch/arm/plat-omap/mcbsp.c
@@ -198,6 +198,170 @@ void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config)
}
EXPORT_SYMBOL(omap_mcbsp_config);
+#ifdef CONFIG_ARCH_OMAP34XX
+/*
+ * omap_mcbsp_set_tx_threshold configures how to deal
+ * with transmit threshold. the threshold value and handler can be
+ * configure in here.
+ */
+void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+
+ if (!cpu_is_omap34xx())
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ OMAP_MCBSP_WRITE(io_base, THRSH2, threshold);
+}
+EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold);
+
+/*
+ * omap_mcbsp_set_rx_threshold configures how to deal
+ * with receive threshold. the threshold value and handler can be
+ * configure in here.
+ */
+void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold)
+{
+ struct omap_mcbsp *mcbsp;
+ void __iomem *io_base;
+
+ if (!cpu_is_omap34xx())
+ return;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+ io_base = mcbsp->io_base;
+
+ OMAP_MCBSP_WRITE(io_base, THRSH1, threshold);
+}
+EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold);
+
+/*
+ * omap_mcbsp_get_max_tx_thres just return the current configured
+ * maximum threshold for transmission
+ */
+u16 omap_mcbsp_get_max_tx_threshold(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ return mcbsp->max_tx_thres;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold);
+
+/*
+ * omap_mcbsp_get_max_rx_thres just return the current configured
+ * maximum threshold for reception
+ */
+u16 omap_mcbsp_get_max_rx_threshold(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ return mcbsp->max_rx_thres;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold);
+
+/*
+ * omap_mcbsp_get_dma_op_mode just return the current configured
+ * operating mode for the mcbsp channel
+ */
+int omap_mcbsp_get_dma_op_mode(unsigned int id)
+{
+ struct omap_mcbsp *mcbsp;
+ int dma_op_mode;
+
+ if (!omap_mcbsp_check_valid_id(id)) {
+ printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1);
+ return -ENODEV;
+ }
+ mcbsp = id_to_mcbsp_ptr(id);
+
+ spin_lock_irq(&mcbsp->lock);
+ dma_op_mode = mcbsp->dma_op_mode;
+ spin_unlock_irq(&mcbsp->lock);
+
+ return dma_op_mode;
+}
+EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode);
+
+static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp)
+{
+ /*
+ * Enable wakup behavior, smart idle and all wakeups
+ * REVISIT: some wakeups may be unnecessary
+ */
+ if (cpu_is_omap34xx()) {
+ u16 syscon;
+
+ syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON);
+ syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03));
+
+ spin_lock_irq(&mcbsp->lock);
+ if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
+ syscon |= (ENAWAKEUP | SIDLEMODE(0x02) |
+ CLOCKACTIVITY(0x02));
+ OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN,
+ XRDYEN | RRDYEN);
+ } else {
+ syscon |= SIDLEMODE(0x01);
+ }
+ spin_unlock_irq(&mcbsp->lock);
+
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+ }
+}
+
+static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp)
+{
+ /*
+ * Disable wakup behavior, smart idle and all wakeups
+ */
+ if (cpu_is_omap34xx()) {
+ u16 syscon;
+
+ syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON);
+ syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03));
+ /*
+ * HW bug workaround - If no_idle mode is taken, we need to
+ * go to smart_idle before going to always_idle, or the
+ * device will not hit retention anymore.
+ */
+ syscon |= SIDLEMODE(0x02);
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+
+ syscon &= ~(SIDLEMODE(0x03));
+ OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon);
+
+ OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN, 0);
+ }
+}
+#else
+static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp) {}
+static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp) {}
+#endif
+
/*
* We can choose between IRQ based or polled IO.
* This needs to be called before omap_mcbsp_request().
@@ -257,6 +421,9 @@ int omap_mcbsp_request(unsigned int id)
clk_enable(mcbsp->iclk);
clk_enable(mcbsp->fclk);
+ /* Do procedure specific to omap34xx arch, if applicable */
+ omap34xx_mcbsp_request(mcbsp);
+
/*
* Make sure that transmitter, receiver and sample-rate generator are
* not running before activating IRQs.
@@ -305,6 +472,9 @@ void omap_mcbsp_free(unsigned int id)
if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(id);
+ /* Do procedure specific to omap34xx arch, if applicable */
+ omap34xx_mcbsp_free(mcbsp);
+
clk_disable(mcbsp->fclk);
clk_disable(mcbsp->iclk);
@@ -328,14 +498,15 @@ void omap_mcbsp_free(unsigned int id)
EXPORT_SYMBOL(omap_mcbsp_free);
/*
- * Here we start the McBSP, by enabling the sample
- * generator, both transmitter and receivers,
- * and the frame sync.
+ * Here we start the McBSP, by enabling transmitter, receiver or both.
+ * If no transmitter or receiver is active prior calling, then sample-rate
+ * generator and frame sync are started.
*/
-void omap_mcbsp_start(unsigned int id)
+void omap_mcbsp_start(unsigned int id, int tx, int rx)
{
struct omap_mcbsp *mcbsp;
void __iomem *io_base;
+ int idle;
u16 w;
if (!omap_mcbsp_check_valid_id(id)) {
@@ -348,32 +519,58 @@ void omap_mcbsp_start(unsigned int id)
mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7;
mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7;
- /* Start the sample generator */
- w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6));
+ idle = !((OMAP_MCBSP_READ(io_base, SPCR2) |
+ OMAP_MCBSP_READ(io_base, SPCR1)) & 1);
+
+ if (idle) {
+ /* Start the sample generator */
+ w = OMAP_MCBSP_READ(io_base, SPCR2);
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6));
+ }
/* Enable transmitter and receiver */
+ tx &= 1;
w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w | 1);
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w | tx);
+ rx &= 1;
w = OMAP_MCBSP_READ(io_base, SPCR1);
- OMAP_MCBSP_WRITE(io_base, SPCR1, w | 1);
+ OMAP_MCBSP_WRITE(io_base, SPCR1, w | rx);
- udelay(100);
+ /*
+ * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec
+ * REVISIT: 100us may give enough time for two CLKSRG, however
+ * due to some unknown PM related, clock gating etc. reason it
+ * is now at 500us.
+ */
+ udelay(500);
- /* Start frame sync */
- w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7));
+ if (idle) {
+ /* Start frame sync */
+ w = OMAP_MCBSP_READ(io_base, SPCR2);
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7));
+ }
+
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ /* Release the transmitter and receiver */
+ w = OMAP_MCBSP_READ(io_base, XCCR);
+ w &= ~(tx ? XDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, XCCR, w);
+ w = OMAP_MCBSP_READ(io_base, RCCR);
+ w &= ~(rx ? RDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, RCCR, w);
+ }
/* Dump McBSP Regs */
omap_mcbsp_dump_reg(id);
}
EXPORT_SYMBOL(omap_mcbsp_start);
-void omap_mcbsp_stop(unsigned int id)
+void omap_mcbsp_stop(unsigned int id, int tx, int rx)
{
struct omap_mcbsp *mcbsp;
void __iomem *io_base;
+ int idle;
u16 w;
if (!omap_mcbsp_check_valid_id(id)) {
@@ -385,16 +582,33 @@ void omap_mcbsp_stop(unsigned int id)
io_base = mcbsp->io_base;
/* Reset transmitter */
+ tx &= 1;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ w = OMAP_MCBSP_READ(io_base, XCCR);
+ w |= (tx ? XDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, XCCR, w);
+ }
w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1));
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~tx);
/* Reset receiver */
+ rx &= 1;
+ if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+ w = OMAP_MCBSP_READ(io_base, RCCR);
+ w |= (tx ? RDISABLE : 0);
+ OMAP_MCBSP_WRITE(io_base, RCCR, w);
+ }
w = OMAP_MCBSP_READ(io_base, SPCR1);
- OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(1));
+ OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~rx);
- /* Reset the sample rate generator */
- w = OMAP_MCBSP_READ(io_base, SPCR2);
- OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6));
+ idle = !((OMAP_MCBSP_READ(io_base, SPCR2) |
+ OMAP_MCBSP_READ(io_base, SPCR1)) & 1);
+
+ if (idle) {
+ /* Reset the sample rate generator */
+ w = OMAP_MCBSP_READ(io_base, SPCR2);
+ OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6));
+ }
}
EXPORT_SYMBOL(omap_mcbsp_stop);
@@ -883,6 +1097,149 @@ void omap_mcbsp_set_spi_mode(unsigned int id,
}
EXPORT_SYMBOL(omap_mcbsp_set_spi_mode);
+#ifdef CONFIG_ARCH_OMAP34XX
+#define max_thres(m) (mcbsp->pdata->buffer_size)
+#define valid_threshold(m, val) ((val) <= max_thres(m))
+#define THRESHOLD_PROP_BUILDER(prop) \
+static ssize_t prop##_show(struct device *dev, \
+ struct device_attribute *attr, char *buf) \
+{ \
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
+ \
+ return sprintf(buf, "%u\n", mcbsp->prop); \
+} \
+ \
+static ssize_t prop##_store(struct device *dev, \
+ struct device_attribute *attr, \
+ const char *buf, size_t size) \
+{ \
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \
+ unsigned long val; \
+ int status; \
+ \
+ status = strict_strtoul(buf, 0, &val); \
+ if (status) \
+ return status; \
+ \
+ if (!valid_threshold(mcbsp, val)) \
+ return -EDOM; \
+ \
+ mcbsp->prop = val; \
+ return size; \
+} \
+ \
+static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store);
+
+THRESHOLD_PROP_BUILDER(max_tx_thres);
+THRESHOLD_PROP_BUILDER(max_rx_thres);
+
+static const char *dma_op_modes[] = {
+ "element", "threshold", "frame",
+};
+
+static ssize_t dma_op_mode_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+ int dma_op_mode, i = 0;
+ ssize_t len = 0;
+ const char * const *s;
+
+ spin_lock_irq(&mcbsp->lock);
+ dma_op_mode = mcbsp->dma_op_mode;
+ spin_unlock_irq(&mcbsp->lock);
+
+ for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) {
+ if (dma_op_mode == i)
+ len += sprintf(buf + len, "[%s] ", *s);
+ else
+ len += sprintf(buf + len, "%s ", *s);
+ }
+ len += sprintf(buf + len, "\n");
+
+ return len;
+}
+
+static ssize_t dma_op_mode_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t size)
+{
+ struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+ const char * const *s;
+ int i = 0;
+
+ for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++)
+ if (sysfs_streq(buf, *s))
+ break;
+
+ if (i == ARRAY_SIZE(dma_op_modes))
+ return -EINVAL;
+
+ spin_lock_irq(&mcbsp->lock);
+ if (!mcbsp->free) {
+ size = -EBUSY;
+ goto unlock;
+ }
+ mcbsp->dma_op_mode = i;
+
+unlock:
+ spin_unlock_irq(&mcbsp->lock);
+
+ return size;
+}
+
+static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store);
+
+static const struct attribute *additional_attrs[] = {
+ &dev_attr_max_tx_thres.attr,
+ &dev_attr_max_rx_thres.attr,
+ &dev_attr_dma_op_mode.attr,
+ NULL,
+};
+
+static const struct attribute_group additional_attr_group = {
+ .attrs = (struct attribute **)additional_attrs,
+};
+
+static inline int __devinit omap_additional_add(struct device *dev)
+{
+ return sysfs_create_group(&dev->kobj, &additional_attr_group);
+}
+
+static inline void __devexit omap_additional_remove(struct device *dev)
+{
+ sysfs_remove_group(&dev->kobj, &additional_attr_group);
+}
+
+static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp)
+{
+ mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT;
+ if (cpu_is_omap34xx()) {
+ mcbsp->max_tx_thres = max_thres(mcbsp);
+ mcbsp->max_rx_thres = max_thres(mcbsp);
+ /*
+ * REVISIT: Set dmap_op_mode to THRESHOLD as default
+ * for mcbsp2 instances.
+ */
+ if (omap_additional_add(mcbsp->dev))
+ dev_warn(mcbsp->dev,
+ "Unable to create additional controls\n");
+ } else {
+ mcbsp->max_tx_thres = -EINVAL;
+ mcbsp->max_rx_thres = -EINVAL;
+ }
+}
+
+static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp)
+{
+ if (cpu_is_omap34xx())
+ omap_additional_remove(mcbsp->dev);
+}
+#else
+static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp) {}
+static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp) {}
+#endif /* CONFIG_ARCH_OMAP34XX */
+
/*
* McBSP1 and McBSP3 are directly mapped on 1610 and 1510.
* 730 has only 2 McBSP, and both of them are MPU peripherals.
@@ -953,6 +1310,10 @@ static int __devinit omap_mcbsp_probe(struct platform_device *pdev)
mcbsp->dev = &pdev->dev;
mcbsp_ptr[id] = mcbsp;
platform_set_drvdata(pdev, mcbsp);
+
+ /* Initialize mcbsp properties for OMAP34XX if needed / applicable */
+ omap34xx_device_init(mcbsp);
+
return 0;
err_fclk:
@@ -976,6 +1337,8 @@ static int __devexit omap_mcbsp_remove(struct platform_device *pdev)
mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(mcbsp->id);
+ omap34xx_device_exit(mcbsp);
+
clk_disable(mcbsp->fclk);
clk_disable(mcbsp->iclk);
clk_put(mcbsp->fclk);
diff --git a/arch/arm/plat-s3c/include/plat/audio-simtec.h b/arch/arm/plat-s3c/include/plat/audio-simtec.h
new file mode 100644
index 0000000..0f440b9
--- /dev/null
+++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h
@@ -0,0 +1,37 @@
+/* arch/arm/plat-s3c/include/plat/audio-simtec.h
+ *
+ * Copyright 2008 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Simtec Audio support.
+*/
+
+/**
+ * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio
+ * @use_mpllin: Select codec clock from MPLLin
+ * @output_cdclk: Need to output CDCLK to the codec
+ * @have_mic: Set if we have a MIC socket
+ * @have_lout: Set if we have a LineOut socket
+ * @amp_gpio: GPIO pin to enable the AMP
+ * @amp_gain: Option GPIO to control AMP gain
+ */
+struct s3c24xx_audio_simtec_pdata {
+ unsigned int use_mpllin:1;
+ unsigned int output_cdclk:1;
+
+ unsigned int have_mic:1;
+ unsigned int have_lout:1;
+
+ int amp_gpio;
+ int amp_gain[2];
+
+ void (*startup)(void);
+};
+
+extern int simtec_audio_add(const char *codec_name,
+ struct s3c24xx_audio_simtec_pdata *pdata);
diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
index 0fad757..07659da 100644
--- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
+++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
@@ -33,6 +33,11 @@
#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1)
#define S3C2412_IISCON_IIS_ACTIVE (1 << 0)
+#define S3C64XX_IISMOD_BLC_16BIT (0 << 13)
+#define S3C64XX_IISMOD_BLC_8BIT (1 << 13)
+#define S3C64XX_IISMOD_BLC_24BIT (2 << 13)
+#define S3C64XX_IISMOD_BLC_MASK (3 << 13)
+
#define S3C64XX_IISMOD_IMS_PCLK (0 << 10)
#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10)
diff --git a/fs/char_dev.c b/fs/char_dev.c
index 7c27a8e..3cbc57f 100644
--- a/fs/char_dev.c
+++ b/fs/char_dev.c
@@ -238,8 +238,10 @@ int alloc_chrdev_region(dev_t *dev, unsigned baseminor, unsigned count,
}
/**
- * register_chrdev() - Register a major number for character devices.
+ * __register_chrdev() - create and register a cdev occupying a range of minors
* @major: major device number or 0 for dynamic allocation
+ * @baseminor: first of the requested range of minor numbers
+ * @count: the number of minor numbers required
* @name: name of this range of devices
* @fops: file operations associated with this devices
*
@@ -255,19 +257,17 @@ int alloc_chrdev_region(dev_t *dev, unsigned baseminor, unsigned count,
* /dev. It only helps to keep track of the different owners of devices. If
* your module name has only one type of devices it's ok to use e.g. the name
* of the module here.
- *
- * This function registers a range of 256 minor numbers. The first minor number
- * is 0.
*/
-int register_chrdev(unsigned int major, const char *name,
- const struct file_operations *fops)
+int __register_chrdev(unsigned int major, unsigned int baseminor,
+ unsigned int count, const char *name,
+ const struct file_operations *fops)
{
struct char_device_struct *cd;
struct cdev *cdev;
char *s;
int err = -ENOMEM;
- cd = __register_chrdev_region(major, 0, 256, name);
+ cd = __register_chrdev_region(major, baseminor, count, name);
if (IS_ERR(cd))
return PTR_ERR(cd);
@@ -281,7 +281,7 @@ int register_chrdev(unsigned int major, const char *name,
for (s = strchr(kobject_name(&cdev->kobj),'/'); s; s = strchr(s, '/'))
*s = '!';
- err = cdev_add(cdev, MKDEV(cd->major, 0), 256);
+ err = cdev_add(cdev, MKDEV(cd->major, baseminor), count);
if (err)
goto out;
@@ -291,7 +291,7 @@ int register_chrdev(unsigned int major, const char *name,
out:
kobject_put(&cdev->kobj);
out2:
- kfree(__unregister_chrdev_region(cd->major, 0, 256));
+ kfree(__unregister_chrdev_region(cd->major, baseminor, count));
return err;
}
@@ -317,10 +317,23 @@ void unregister_chrdev_region(dev_t from, unsigned count)
}
}
-void unregister_chrdev(unsigned int major, const char *name)
+/**
+ * __unregister_chrdev - unregister and destroy a cdev
+ * @major: major device number
+ * @baseminor: first of the range of minor numbers
+ * @count: the number of minor numbers this cdev is occupying
+ * @name: name of this range of devices
+ *
+ * Unregister and destroy the cdev occupying the region described by
+ * @major, @baseminor and @count. This function undoes what
+ * __register_chrdev() did.
+ */
+void __unregister_chrdev(unsigned int major, unsigned int baseminor,
+ unsigned int count, const char *name)
{
struct char_device_struct *cd;
- cd = __unregister_chrdev_region(major, 0, 256);
+
+ cd = __unregister_chrdev_region(major, baseminor, count);
if (cd && cd->cdev)
cdev_del(cd->cdev);
kfree(cd);
@@ -569,6 +582,6 @@ EXPORT_SYMBOL(cdev_alloc);
EXPORT_SYMBOL(cdev_del);
EXPORT_SYMBOL(cdev_add);
EXPORT_SYMBOL(cdev_index);
-EXPORT_SYMBOL(register_chrdev);
-EXPORT_SYMBOL(unregister_chrdev);
+EXPORT_SYMBOL(__register_chrdev);
+EXPORT_SYMBOL(__unregister_chrdev);
EXPORT_SYMBOL(directly_mappable_cdev_bdi);
diff --git a/include/linux/fs.h b/include/linux/fs.h
index 26da98f..a79f483 100644
--- a/include/linux/fs.h
+++ b/include/linux/fs.h
@@ -1997,12 +1997,25 @@ extern void bd_release_from_disk(struct block_device *, struct gendisk *);
#define CHRDEV_MAJOR_HASH_SIZE 255
extern int alloc_chrdev_region(dev_t *, unsigned, unsigned, const char *);
extern int register_chrdev_region(dev_t, unsigned, const char *);
-extern int register_chrdev(unsigned int, const char *,
- const struct file_operations *);
-extern void unregister_chrdev(unsigned int, const char *);
+extern int __register_chrdev(unsigned int major, unsigned int baseminor,
+ unsigned int count, const char *name,
+ const struct file_operations *fops);
+extern void __unregister_chrdev(unsigned int major, unsigned int baseminor,
+ unsigned int count, const char *name);
extern void unregister_chrdev_region(dev_t, unsigned);
extern void chrdev_show(struct seq_file *,off_t);
+static inline int register_chrdev(unsigned int major, const char *name,
+ const struct file_operations *fops)
+{
+ return __register_chrdev(major, 0, 256, name, fops);
+}
+
+static inline void unregister_chrdev(unsigned int major, const char *name)
+{
+ __unregister_chrdev(major, 0, 256, name);
+}
+
/* fs/block_dev.c */
#define BDEVNAME_SIZE 32 /* Largest string for a blockdev identifier */
#define BDEVT_SIZE 10 /* Largest string for MAJ:MIN for blkdev */
diff --git a/include/linux/tty.h b/include/linux/tty.h
index e8c6c91..0d3974f 100644
--- a/include/linux/tty.h
+++ b/include/linux/tty.h
@@ -23,7 +23,7 @@
*/
#define NR_UNIX98_PTY_DEFAULT 4096 /* Default maximum for Unix98 ptys */
#define NR_UNIX98_PTY_MAX (1 << MINORBITS) /* Absolute limit */
-#define NR_LDISCS 19
+#define NR_LDISCS 20
/* line disciplines */
#define N_TTY 0
@@ -47,6 +47,8 @@
#define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */
#define N_PPS 18 /* Pulse per Second */
+#define N_V253 19 /* Codec control over voice modem */
+
/*
* This character is the same as _POSIX_VDISABLE: it cannot be used as
* a c_cc[] character, but indicates that a particular special character
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 251fc1c..3dae3f7 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -32,6 +32,9 @@
#include "control.h"
#include "info.h"
+/* maximum number of devices on the AC97 bus */
+#define AC97_BUS_MAX_DEVICES 4
+
/*
* AC'97 codec registers
*/
@@ -642,4 +645,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
/* ad hoc AC97 device driver access */
extern struct bus_type ac97_bus_type;
+/* AC97 platform_data adding function */
+static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
+{
+ ac97->dev.platform_data = data;
+}
+
#endif /* __SOUND_AC97_CODEC_H */
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 82aed3f..1f57bb9 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -138,7 +138,7 @@ struct snd_hwdep_dsp_image {
* *
*****************************************************************************/
-#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9)
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
diff --git a/include/sound/core.h b/include/sound/core.h
index 309cb96..a61499c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -93,15 +93,6 @@ struct snd_device {
#define snd_device(n) list_entry(n, struct snd_device, list)
-/* monitor files for graceful shutdown (hotplug) */
-
-struct snd_monitor_file {
- struct file *file;
- const struct file_operations *disconnected_f_op;
- struct list_head shutdown_list; /* still need to shutdown */
- struct list_head list; /* link of monitor files */
-};
-
/* main structure for soundcard */
struct snd_card {
@@ -311,9 +302,7 @@ int snd_component_add(struct snd_card *card, const char *component);
int snd_card_file_add(struct snd_card *card, struct file *file);
int snd_card_file_remove(struct snd_card *card, struct file *file);
-#ifndef snd_card_set_dev
#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr))
-#endif
/* device.c */
@@ -340,18 +329,17 @@ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size);
struct resource;
void release_and_free_resource(struct resource *res);
-#ifdef CONFIG_SND_VERBOSE_PRINTK
-void snd_verbose_printk(const char *file, int line, const char *format, ...)
- __attribute__ ((format (printf, 3, 4)));
-#endif
-#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK)
-void snd_verbose_printd(const char *file, int line, const char *format, ...)
- __attribute__ ((format (printf, 3, 4)));
-#endif
-
/* --- */
-#ifdef CONFIG_SND_VERBOSE_PRINTK
+#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+void __snd_printk(unsigned int level, const char *file, int line,
+ const char *format, ...)
+ __attribute__ ((format (printf, 4, 5)));
+#else
+#define __snd_printk(level, file, line, format, args...) \
+ printk(format, ##args)
+#endif
+
/**
* snd_printk - printk wrapper
* @fmt: format string
@@ -360,15 +348,9 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
* when configured with CONFIG_SND_VERBOSE_PRINTK.
*/
#define snd_printk(fmt, args...) \
- snd_verbose_printk(__FILE__, __LINE__, fmt ,##args)
-#else
-#define snd_printk(fmt, args...) \
- printk(fmt ,##args)
-#endif
+ __snd_printk(0, __FILE__, __LINE__, fmt, ##args)
#ifdef CONFIG_SND_DEBUG
-
-#ifdef CONFIG_SND_VERBOSE_PRINTK
/**
* snd_printd - debug printk
* @fmt: format string
@@ -377,11 +359,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
* Ignored when CONFIG_SND_DEBUG is not set.
*/
#define snd_printd(fmt, args...) \
- snd_verbose_printd(__FILE__, __LINE__, fmt ,##args)
-#else
-#define snd_printd(fmt, args...) \
- printk(fmt ,##args)
-#endif
+ __snd_printk(1, __FILE__, __LINE__, fmt, ##args)
/**
* snd_BUG - give a BUG warning message and stack trace
@@ -428,9 +406,10 @@ static inline int __snd_bug_on(int cond)
* Works like snd_printk() for debugging purposes.
* Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
*/
-#define snd_printdd(format, args...) snd_printk(format, ##args)
+#define snd_printdd(format, args...) \
+ __snd_printk(2, __FILE__, __LINE__, format, ##args)
#else
-#define snd_printdd(format, args...) /* nothing */
+#define snd_printdd(format, args...) do { } while (0)
#endif
@@ -438,12 +417,10 @@ static inline int __snd_bug_on(int cond)
/* for easier backward-porting */
#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE)
-#ifndef gameport_set_dev_parent
#define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev))
#define gameport_set_port_data(gp,r) ((gp)->port_data = (r))
#define gameport_get_port_data(gp) (gp)->port_data
#endif
-#endif
/* PCI quirk list helper */
struct snd_pci_quirk {
diff --git a/include/sound/info.h b/include/sound/info.h
index 7c2ee1a2..112e894 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -110,13 +110,13 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
-int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \
+int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) \
__attribute__ ((format (printf, 2, 3)));
int snd_info_init(void);
int snd_info_done(void);
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
-char *snd_info_get_str(char *dest, char *src, int len);
+const char *snd_info_get_str(char *dest, const char *src, int len);
struct snd_info_entry *snd_info_create_module_entry(struct module *module,
const char *name,
struct snd_info_entry *parent);
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 7ccce94..c425062 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -47,7 +47,11 @@ struct snd_dma_device {
#define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */
#define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */
#define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */
+#ifdef CONFIG_SND_DMA_SGBUF
#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */
+#else
+#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */
+#endif
/*
* info for buffer allocation
@@ -60,6 +64,7 @@ struct snd_dma_buffer {
void *private_data; /* private for allocator; don't touch */
};
+#ifdef CONFIG_SND_DMA_SGBUF
/*
* Scatter-Gather generic device pages
*/
@@ -107,6 +112,7 @@ static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset)
{
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
}
+#endif /* CONFIG_SND_DMA_SGBUF */
/* allocate/release a buffer */
int snd_dma_alloc_pages(int type, struct device *dev, size_t size,
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2389352..de6d981 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -902,6 +902,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size);
int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream);
+#ifdef CONFIG_SND_DMA_SGBUF
/*
* SG-buffer handling
*/
@@ -927,6 +928,28 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream,
unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
unsigned int ofs, unsigned int size);
+#else /* !SND_DMA_SGBUF */
+/*
+ * fake using a continuous buffer
+ */
+static inline dma_addr_t
+snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return substream->runtime->dma_addr + ofs;
+}
+
+static inline void *
+snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return substream->runtime->dma_area + ofs;
+}
+
+#define snd_pcm_sgbuf_ops_page NULL
+
+#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size)
+
+#endif /* SND_DMA_SGBUF */
+
/* handle mmap counter - PCM mmap callback should handle this counter properly */
static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area)
{
@@ -965,4 +988,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
#define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime)
+const char *snd_pcm_format_name(snd_pcm_format_t format);
+
#endif /* __SOUND_PCM_H */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
new file mode 100644
index 0000000..c022736
--- /dev/null
+++ b/include/sound/sh_fsi.h
@@ -0,0 +1,83 @@
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/* flags format
+
+ * 0xABCDEEFF
+ *
+ * A: channel size for TDM (input)
+ * B: channel size for TDM (ooutput)
+ * C: inversion
+ * D: mode
+ * E: input format
+ * F: output format
+ */
+
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+/* TDM channel */
+#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28)
+#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24)
+
+#define SH_FSI_CH_IMASK 0xF0000000
+#define SH_FSI_CH_OMASK 0x0F000000
+#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28)
+#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24)
+
+/* clock inversion */
+#define SH_FSI_INVERSION_MASK 0x00F00000
+#define SH_FSI_LRM_INV (1 << 20)
+#define SH_FSI_BRM_INV (1 << 21)
+#define SH_FSI_LRS_INV (1 << 22)
+#define SH_FSI_BRS_INV (1 << 23)
+
+/* mode */
+#define SH_FSI_MODE_MASK 0x000F0000
+#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */
+#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */
+
+/* DI format */
+#define SH_FSI_FMT_MASK 0x000000FF
+#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8)
+#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0)
+#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK)
+#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK)
+
+#define SH_FSI_FMT_MONO (1 << 0)
+#define SH_FSI_FMT_MONO_DELAY (1 << 1)
+#define SH_FSI_FMT_PCM (1 << 2)
+#define SH_FSI_FMT_I2S (1 << 3)
+#define SH_FSI_FMT_TDM (1 << 4)
+#define SH_FSI_FMT_TDM_DELAY (1 << 5)
+
+#define SH_FSI_IFMT_TDM_CH(x) \
+ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x))
+#define SH_FSI_IFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x))
+
+#define SH_FSI_OFMT_TDM_CH(x) \
+ (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x))
+#define SH_FSI_OFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x))
+
+struct sh_fsi_platform_info {
+ unsigned long porta_flags;
+ unsigned long portb_flags;
+};
+
+extern struct snd_soc_dai fsi_soc_dai[2];
+extern struct snd_soc_platform fsi_soc_platform;
+
+#endif /* __SOUND_FSI_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 352d7ee..97ca9af 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -27,8 +27,8 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */
-#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */
+#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
/* left and right justified also known as MSB and LSB respectively */
@@ -38,7 +38,7 @@ struct snd_pcm_substream;
/*
* DAI Clock gating.
*
- * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * DAI bit clocks can be be gated (disabled) when the DAI is not
* sending or receiving PCM data in a frame. This can be used to save power.
*/
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
@@ -51,21 +51,21 @@ struct snd_pcm_substream;
* format.
*/
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
-#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */
-#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */
-#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */
+#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
/*
* DAI hardware clock masters.
*
* This is wrt the codec, the inverse is true for the interface
- * i.e. if the codec is clk and frm master then the interface is
+ * i.e. if the codec is clk and FRM master then the interface is
* clk and frame slave.
*/
-#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */
-#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */
+#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
@@ -78,7 +78,13 @@ struct snd_pcm_substream;
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
-#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+ SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S20_3BE |\
+ SNDRV_PCM_FMTBIT_S24_3LE |\
+ SNDRV_PCM_FMTBIT_S24_3BE |\
SNDRV_PCM_FMTBIT_S32_LE |\
SNDRV_PCM_FMTBIT_S32_BE)
@@ -106,7 +112,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
- unsigned int mask, int slots);
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
@@ -116,12 +122,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
/*
* Digital Audio Interface.
*
- * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
- * operations an capabilities. Codec and platfom drivers will register a this
+ * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
+ * operations and capabilities. Codec and platform drivers will register this
* structure for every DAI they have.
*
* This structure covers the clocking, formating and ALSA operations for each
- * interface a
+ * interface.
*/
struct snd_soc_dai_ops {
/*
@@ -140,7 +146,8 @@ struct snd_soc_dai_ops {
*/
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
- unsigned int mask, int slots);
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
@@ -179,6 +186,7 @@ struct snd_soc_dai {
int ac97_control;
struct device *dev;
+ void *ac97_pdata; /* platform_data for the ac97 codec */
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ec8a45f..c1410e3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -137,6 +137,12 @@
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
/* stream domain */
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
@@ -279,9 +285,11 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
+void snd_soc_dapm_shutdown(struct snd_soc_device *socdev);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
@@ -311,6 +319,8 @@ enum snd_soc_dapm_type {
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
+ snd_soc_dapm_aif_in, /* audio interface input */
+ snd_soc_dapm_aif_out, /* audio interface output */
};
/*
diff --git a/include/sound/soc.h b/include/sound/soc.h
index cf6111d..475cb7e 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -135,6 +135,28 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+ .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .max = xmax, .invert = xinvert} }
#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_bool_ext, \
@@ -183,14 +205,28 @@ struct snd_soc_jack_gpio;
#endif
typedef int (*hw_write_t)(void *,const char* ,int);
-typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
+enum snd_soc_control_type {
+ SND_SOC_CUSTOM,
+ SND_SOC_I2C,
+ SND_SOC_SPI,
+};
+
int snd_soc_register_platform(struct snd_soc_platform *platform);
void snd_soc_unregister_platform(struct snd_soc_platform *platform);
int snd_soc_register_codec(struct snd_soc_codec *codec);
void snd_soc_unregister_codec(struct snd_soc_codec *codec);
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg);
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control);
+
+#ifdef CONFIG_PM
+int snd_soc_suspend_device(struct device *dev);
+int snd_soc_resume_device(struct device *dev);
+#endif
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
@@ -216,9 +252,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value);
+ unsigned int mask, unsigned int value);
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value);
+ unsigned int mask, unsigned int value);
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num);
@@ -356,8 +392,10 @@ struct snd_soc_codec {
int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
int (*display_register)(struct snd_soc_codec *, char *,
size_t, unsigned int);
+ int (*volatile_register)(unsigned int);
+ int (*readable_register)(unsigned int);
hw_write_t hw_write;
- hw_read_t hw_read;
+ unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
void *reg_cache;
short reg_cache_size;
short reg_cache_step;
@@ -369,8 +407,6 @@ struct snd_soc_codec {
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
- struct list_head up_list;
- struct list_head down_list;
/* codec DAI's */
struct snd_soc_dai *dai;
@@ -379,6 +415,7 @@ struct snd_soc_codec {
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
+ struct dentry *debugfs_dapm;
#endif
};
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index d136ea2..9fd5b19 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -35,6 +35,8 @@
#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */
#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */
#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */
+#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */
+#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */
#define TLV_DB_SCALE_ITEM(min, step, mute) \
SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \
@@ -42,6 +44,18 @@
#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
+/* dB scale specified with min/max values instead of step */
+#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \
+ (min_dB), (max_dB)
+#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \
+ (min_dB), (max_dB)
+#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
+#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
+
/* linear volume between min_dB and max_dB (.01dB unit) */
#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \
diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h
new file mode 100644
index 0000000..381319c
--- /dev/null
+++ b/include/sound/uda1380.h
@@ -0,0 +1,22 @@
+/*
+ * UDA1380 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __UDA1380_H
+#define __UDA1380_H
+
+struct uda1380_platform_data {
+ int gpio_power;
+ int gpio_reset;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#endif /* __UDA1380_H */
diff --git a/include/sound/version.h b/include/sound/version.h
index 456f135..2293914 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.20"
+#define CONFIG_SND_VERSION "1.0.21"
#define CONFIG_SND_DATE ""
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
new file mode 100644
index 0000000..9c661f2
--- /dev/null
+++ b/include/sound/wm8993.h
@@ -0,0 +1,44 @@
+/*
+ * linux/sound/wm8993.h -- Platform data for WM8993
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM8993_H
+#define __LINUX_SND_WM8993_H
+
+/* Note that EQ1 only contains the enable/disable bit so will be
+ ignored but is included for simplicity.
+ */
+struct wm8993_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[24];
+};
+
+struct wm8993_platform_data {
+ struct wm8993_retune_mobile_setting *retune_configs;
+ int num_retune_configs;
+
+ /* LINEOUT can be differential or single ended */
+ unsigned int lineout1_diff:1;
+ unsigned int lineout2_diff:1;
+
+ /* Common mode feedback */
+ unsigned int lineout1fb:1;
+ unsigned int lineout2fb:1;
+
+ /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+ unsigned int micbias1_lvl:1;
+ unsigned int micbias2_lvl:1;
+
+ /* Jack detect threashold levels, see datasheet for values */
+ unsigned int jd_scthr:2;
+ unsigned int jd_thr:2;
+};
+
+#endif
diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h
index 05ead66..444cd6b 100644
--- a/include/sound/ymfpci.h
+++ b/include/sound/ymfpci.h
@@ -331,6 +331,7 @@ struct snd_ymfpci {
struct snd_ac97 *ac97;
struct snd_rawmidi *rawmidi;
struct snd_timer *timer;
+ unsigned int timer_ticks;
struct pci_dev *pci;
struct snd_card *card;
diff --git a/sound/Kconfig b/sound/Kconfig
index 1eceb85..439e15c 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -32,6 +32,34 @@ config SOUND_OSS_CORE
bool
default n
+config SOUND_OSS_CORE_PRECLAIM
+ bool "Preclaim OSS device numbers"
+ depends on SOUND_OSS_CORE
+ default y
+ help
+ With this option enabled, the kernel will claim all OSS device
+ numbers if any OSS support (native or emulation) is enabled
+ whether the respective module is loaded or not and try to load the
+ appropriate module using sound-slot/service-* and char-major-*
+ module aliases when one of the device numbers is opened. With
+ this option disabled, kernel will only claim actually in-use
+ device numbers and opening a missing device will generate only the
+ standard char-major-* aliases.
+
+ The only visible difference is use of additional module aliases
+ and whether OSS sound devices appear multiple times in
+ /proc/devices. sound-slot/service-* module aliases are scheduled
+ to be removed (ie. PRECLAIM won't be available) and this option is
+ to make the transition easier. This option can be overridden
+ during boot using the kernel parameter soundcore.preclaim_oss.
+
+ Disabling this allows alternative OSS implementations.
+
+ Please read Documentation/feature-removal-schedule.txt for
+ details.
+
+ If unusre, say Y.
+
source "sound/oss/dmasound/Kconfig"
if !M68K
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index c570ebd..4e34d19 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -170,6 +170,13 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
struct snd_ac97_bus *ac97_bus;
struct snd_ac97_template ac97_template;
int ret;
+ pxa2xx_audio_ops_t *pdata = dev->dev.platform_data;
+
+ if (dev->id >= 0) {
+ dev_err(&dev->dev, "PXA2xx has only one AC97 port.\n");
+ ret = -ENXIO;
+ goto err_dev;
+ }
ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
THIS_MODULE, 0, &card);
@@ -200,6 +207,8 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
snprintf(card->longname, sizeof(card->longname),
"%s (%s)", dev->dev.driver->name, card->mixername);
+ if (pdata && pdata->codec_pdata[0])
+ snd_ac97_dev_add_pdata(ac97_bus->codec[0], pdata->codec_pdata[0]);
snd_card_set_dev(card, &dev->dev);
ret = snd_card_register(card);
if (ret == 0) {
@@ -212,6 +221,7 @@ err_remove:
err:
if (card)
snd_card_free(card);
+err_dev:
return ret;
}
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 6205f37..743ac6a 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -136,6 +136,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
{
struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ if (!prtd || !prtd->params)
+ return 0;
+
DCSR(prtd->dma_ch) &= ~DCSR_RUN;
DCSR(prtd->dma_ch) = 0;
DCMD(prtd->dma_ch) = 0;
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6061fb5..c15682a 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -206,4 +206,8 @@ config SND_PCM_XRUN_DEBUG
config SND_VMASTER
bool
+config SND_DMA_SGBUF
+ def_bool y
+ depends on X86
+
source "sound/core/seq/Kconfig"
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 4229052..350a08d 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,7 +13,7 @@ snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
pcm_memory.o
snd-page-alloc-y := memalloc.o
-snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
+snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
snd-rawmidi-objs := rawmidi.o
snd-timer-objs := timer.o
diff --git a/sound/core/control.c b/sound/core/control.c
index 17b8d47..a8b7fab 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -414,7 +414,7 @@ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id)
EXPORT_SYMBOL(snd_ctl_remove_id);
/**
- * snd_ctl_remove_unlocked_id - remove the unlocked control of the given id and release it
+ * snd_ctl_remove_user_ctl - remove and release the unlocked user control
* @file: active control handle
* @id: the control id to remove
*
@@ -423,8 +423,8 @@ EXPORT_SYMBOL(snd_ctl_remove_id);
*
* Returns 0 if successful, or a negative error code on failure.
*/
-static int snd_ctl_remove_unlocked_id(struct snd_ctl_file * file,
- struct snd_ctl_elem_id *id)
+static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file,
+ struct snd_ctl_elem_id *id)
{
struct snd_card *card = file->card;
struct snd_kcontrol *kctl;
@@ -433,15 +433,23 @@ static int snd_ctl_remove_unlocked_id(struct snd_ctl_file * file,
down_write(&card->controls_rwsem);
kctl = snd_ctl_find_id(card, id);
if (kctl == NULL) {
- up_write(&card->controls_rwsem);
- return -ENOENT;
+ ret = -ENOENT;
+ goto error;
+ }
+ if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) {
+ ret = -EINVAL;
+ goto error;
}
for (idx = 0; idx < kctl->count; idx++)
if (kctl->vd[idx].owner != NULL && kctl->vd[idx].owner != file) {
- up_write(&card->controls_rwsem);
- return -EBUSY;
+ ret = -EBUSY;
+ goto error;
}
ret = snd_ctl_remove(card, kctl);
+ if (ret < 0)
+ goto error;
+ card->user_ctl_count--;
+error:
up_write(&card->controls_rwsem);
return ret;
}
@@ -951,7 +959,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
if (card->user_ctl_count >= MAX_USER_CONTROLS)
return -ENOMEM;
- if (info->count > 1024)
+ if (info->count < 1)
return -EINVAL;
access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
(info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
@@ -1052,18 +1060,10 @@ static int snd_ctl_elem_remove(struct snd_ctl_file *file,
struct snd_ctl_elem_id __user *_id)
{
struct snd_ctl_elem_id id;
- int err;
if (copy_from_user(&id, _id, sizeof(id)))
return -EFAULT;
- err = snd_ctl_remove_unlocked_id(file, &id);
- if (! err) {
- struct snd_card *card = file->card;
- down_write(&card->controls_rwsem);
- card->user_ctl_count--;
- up_write(&card->controls_rwsem);
- }
- return err;
+ return snd_ctl_remove_user_ctl(file, &id);
}
static int snd_ctl_subscribe_events(struct snd_ctl_file *file, int __user *ptr)
diff --git a/sound/core/info.c b/sound/core/info.c
index 35df614..d749a0d 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -88,12 +88,10 @@ static int resize_info_buffer(struct snd_info_buffer *buffer,
char *nbuf;
nsize = PAGE_ALIGN(nsize);
- nbuf = kmalloc(nsize, GFP_KERNEL);
+ nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL);
if (! nbuf)
return -ENOMEM;
- memcpy(nbuf, buffer->buffer, buffer->len);
- kfree(buffer->buffer);
buffer->buffer = nbuf;
buffer->len = nsize;
return 0;
@@ -108,7 +106,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer,
*
* Returns the size of output string.
*/
-int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...)
+int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...)
{
va_list args;
int len, res;
@@ -727,7 +725,7 @@ EXPORT_SYMBOL(snd_info_get_line);
* Returns the updated pointer of the original string so that
* it can be used for the next call.
*/
-char *snd_info_get_str(char *dest, char *src, int len)
+const char *snd_info_get_str(char *dest, const char *src, int len)
{
int c;
diff --git a/sound/core/init.c b/sound/core/init.c
index d5d40d7..ec4a50c 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -31,6 +31,14 @@
#include <sound/control.h>
#include <sound/info.h>
+/* monitor files for graceful shutdown (hotplug) */
+struct snd_monitor_file {
+ struct file *file;
+ const struct file_operations *disconnected_f_op;
+ struct list_head shutdown_list; /* still need to shutdown */
+ struct list_head list; /* link of monitor files */
+};
+
static DEFINE_SPINLOCK(shutdown_lock);
static LIST_HEAD(shutdown_files);
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 1b3534d..9e92441 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -199,6 +199,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
case SNDRV_DMA_TYPE_DEV:
dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
break;
+#endif
+#ifdef CONFIG_SND_DMA_SGBUF
case SNDRV_DMA_TYPE_DEV_SG:
snd_malloc_sgbuf_pages(device, size, dmab, NULL);
break;
@@ -269,6 +271,8 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
case SNDRV_DMA_TYPE_DEV:
snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
break;
+#endif
+#ifdef CONFIG_SND_DMA_SGBUF
case SNDRV_DMA_TYPE_DEV_SG:
snd_free_sgbuf_pages(dmab);
break;
diff --git a/sound/core/misc.c b/sound/core/misc.c
index a9710e0..23a032c 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -24,6 +24,20 @@
#include <linux/ioport.h>
#include <sound/core.h>
+#ifdef CONFIG_SND_DEBUG
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+#define DEFAULT_DEBUG_LEVEL 2
+#else
+#define DEFAULT_DEBUG_LEVEL 1
+#endif
+
+static int debug = DEFAULT_DEBUG_LEVEL;
+module_param(debug, int, 0644);
+MODULE_PARM_DESC(debug, "Debug level (0 = disable)");
+
+#endif /* CONFIG_SND_DEBUG */
+
void release_and_free_resource(struct resource *res)
{
if (res) {
@@ -35,46 +49,53 @@ void release_and_free_resource(struct resource *res)
EXPORT_SYMBOL(release_and_free_resource);
#ifdef CONFIG_SND_VERBOSE_PRINTK
-void snd_verbose_printk(const char *file, int line, const char *format, ...)
+/* strip the leading path if the given path is absolute */
+static const char *sanity_file_name(const char *path)
{
- va_list args;
-
- if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') {
- char tmp[] = "<0>";
+ if (*path == '/')
+ return strrchr(path, '/') + 1;
+ else
+ return path;
+}
+
+/* print file and line with a certain printk prefix */
+static int print_snd_pfx(unsigned int level, const char *path, int line,
+ const char *format)
+{
+ const char *file = sanity_file_name(path);
+ char tmp[] = "<0>";
+ const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT;
+ int ret = 0;
+
+ if (format[0] == '<' && format[2] == '>') {
tmp[1] = format[1];
- printk("%sALSA %s:%d: ", tmp, file, line);
- format += 3;
- } else {
- printk("ALSA %s:%d: ", file, line);
+ pfx = tmp;
+ ret = 1;
}
- va_start(args, format);
- vprintk(format, args);
- va_end(args);
+ printk("%sALSA %s:%d: ", pfx, file, line);
+ return ret;
}
-
-EXPORT_SYMBOL(snd_verbose_printk);
+#else
+#define print_snd_pfx(level, path, line, format) 0
#endif
-#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK)
-void snd_verbose_printd(const char *file, int line, const char *format, ...)
+#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+void __snd_printk(unsigned int level, const char *path, int line,
+ const char *format, ...)
{
va_list args;
- if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') {
- char tmp[] = "<0>";
- tmp[1] = format[1];
- printk("%sALSA %s:%d: ", tmp, file, line);
- format += 3;
- } else {
- printk(KERN_DEBUG "ALSA %s:%d: ", file, line);
- }
+#ifdef CONFIG_SND_DEBUG
+ if (debug < level)
+ return;
+#endif
va_start(args, format);
+ if (print_snd_pfx(level, path, line, format))
+ format += 3; /* skip the printk level-prefix */
vprintk(format, args);
va_end(args);
-
}
-
-EXPORT_SYMBOL(snd_verbose_printd);
+EXPORT_SYMBOL_GPL(__snd_printk);
#endif
#ifdef CONFIG_PCI
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 5dcd8a5..7724238 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -1154,7 +1154,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_mixer_oss *mixer = entry->private_data;
- char line[128], str[32], idxstr[16], *cptr;
+ char line[128], str[32], idxstr[16];
+ const char *cptr;
int ch, idx;
struct snd_mixer_oss_assign_table *tbl;
struct slot *slot;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index dbe406b..d9c9635 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1043,10 +1043,15 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream)
runtime->oss.channels = params_channels(params);
runtime->oss.rate = params_rate(params);
- runtime->oss.params = 0;
- runtime->oss.prepare = 1;
vfree(runtime->oss.buffer);
runtime->oss.buffer = vmalloc(runtime->oss.period_bytes);
+ if (!runtime->oss.buffer) {
+ err = -ENOMEM;
+ goto failure;
+ }
+
+ runtime->oss.params = 0;
+ runtime->oss.prepare = 1;
runtime->oss.buffer_used = 0;
if (runtime->dma_area)
snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));
@@ -2836,7 +2841,8 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_pcm_str *pstr = entry->private_data;
- char line[128], str[32], task_name[32], *ptr;
+ char line[128], str[32], task_name[32];
+ const char *ptr;
int idx1;
struct snd_pcm_oss_setup *setup, *setup1, template;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 145931a..0c14401 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -162,18 +162,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card,
return -ENOIOCTLCMD;
}
-#ifdef CONFIG_SND_VERBOSE_PROCFS
-
-#define STATE(v) [SNDRV_PCM_STATE_##v] = #v
-#define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v
-#define READY(v) [SNDRV_PCM_READY_##v] = #v
-#define XRUN(v) [SNDRV_PCM_XRUN_##v] = #v
-#define SILENCE(v) [SNDRV_PCM_SILENCE_##v] = #v
-#define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v
-#define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v
-#define START(v) [SNDRV_PCM_START_##v] = #v
#define FORMAT(v) [SNDRV_PCM_FORMAT_##v] = #v
-#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v
static char *snd_pcm_format_names[] = {
FORMAT(S8),
@@ -216,10 +205,23 @@ static char *snd_pcm_format_names[] = {
FORMAT(U18_3BE),
};
-static const char *snd_pcm_format_name(snd_pcm_format_t format)
+const char *snd_pcm_format_name(snd_pcm_format_t format)
{
return snd_pcm_format_names[format];
}
+EXPORT_SYMBOL_GPL(snd_pcm_format_name);
+
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+
+#define STATE(v) [SNDRV_PCM_STATE_##v] = #v
+#define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v
+#define READY(v) [SNDRV_PCM_READY_##v] = #v
+#define XRUN(v) [SNDRV_PCM_XRUN_##v] = #v
+#define SILENCE(v) [SNDRV_PCM_SILENCE_##v] = #v
+#define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v
+#define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v
+#define START(v) [SNDRV_PCM_START_##v] = #v
+#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v
static char *snd_pcm_stream_names[] = {
STREAM(PLAYBACK),
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 9db60d8..30f4108 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -197,12 +197,16 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream,
avail = snd_pcm_capture_avail(runtime);
if (avail > runtime->avail_max)
runtime->avail_max = avail;
- if (avail >= runtime->stop_threshold) {
- if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
+ if (avail >= runtime->buffer_size) {
snd_pcm_drain_done(substream);
- else
+ return -EPIPE;
+ }
+ } else {
+ if (avail >= runtime->stop_threshold) {
xrun(substream);
- return -EPIPE;
+ return -EPIPE;
+ }
}
if (avail >= runtime->control->avail_min)
wake_up(&runtime->sleep);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index a6d4280..caa7796 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -304,6 +304,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all);
+#ifdef CONFIG_SND_DMA_SGBUF
/**
* snd_pcm_sgbuf_ops_page - get the page struct at the given offset
* @substream: the pcm substream instance
@@ -349,6 +350,7 @@ unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
return size;
}
EXPORT_SYMBOL(snd_pcm_sgbuf_get_chunk_size);
+#endif /* CONFIG_SND_DMA_SGBUF */
/**
* snd_pcm_lib_malloc_pages - allocate the DMA buffer
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index ac2150e..59e5fbe 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1343,8 +1343,6 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
{
- if (substream->f_flags & O_NONBLOCK)
- return -EAGAIN;
substream->runtime->trigger_master = substream;
return 0;
}
@@ -1392,7 +1390,6 @@ static struct action_ops snd_pcm_action_drain_init = {
struct drain_rec {
struct snd_pcm_substream *substream;
wait_queue_t wait;
- snd_pcm_uframes_t stop_threshold;
};
static int snd_pcm_drop(struct snd_pcm_substream *substream);
@@ -1404,13 +1401,15 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream);
* After this call, all streams are supposed to be either SETUP or DRAINING
* (capture only) state.
*/
-static int snd_pcm_drain(struct snd_pcm_substream *substream)
+static int snd_pcm_drain(struct snd_pcm_substream *substream,
+ struct file *file)
{
struct snd_card *card;
struct snd_pcm_runtime *runtime;
struct snd_pcm_substream *s;
int result = 0;
int i, num_drecs;
+ int nonblock = 0;
struct drain_rec *drec, drec_tmp, *d;
card = substream->pcm->card;
@@ -1428,6 +1427,15 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
}
}
+ if (file) {
+ if (file->f_flags & O_NONBLOCK)
+ nonblock = 1;
+ } else if (substream->f_flags & O_NONBLOCK)
+ nonblock = 1;
+
+ if (nonblock)
+ goto lock; /* no need to allocate waitqueues */
+
/* allocate temporary record for drain sync */
down_read(&snd_pcm_link_rwsem);
if (snd_pcm_stream_linked(substream)) {
@@ -1449,16 +1457,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
d->substream = s;
init_waitqueue_entry(&d->wait, current);
add_wait_queue(&runtime->sleep, &d->wait);
- /* stop_threshold fixup to avoid endless loop when
- * stop_threshold > buffer_size
- */
- d->stop_threshold = runtime->stop_threshold;
- if (runtime->stop_threshold > runtime->buffer_size)
- runtime->stop_threshold = runtime->buffer_size;
}
}
up_read(&snd_pcm_link_rwsem);
+ lock:
snd_pcm_stream_lock_irq(substream);
/* resume pause */
if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED)
@@ -1466,9 +1469,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
/* pre-start/stop - all running streams are changed to DRAINING state */
result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0);
- if (result < 0) {
- snd_pcm_stream_unlock_irq(substream);
- goto _error;
+ if (result < 0)
+ goto unlock;
+ /* in non-blocking, we don't wait in ioctl but let caller poll */
+ if (nonblock) {
+ result = -EAGAIN;
+ goto unlock;
}
for (;;) {
@@ -1504,18 +1510,18 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
}
}
+ unlock:
snd_pcm_stream_unlock_irq(substream);
- _error:
- for (i = 0; i < num_drecs; i++) {
- d = &drec[i];
- runtime = d->substream->runtime;
- remove_wait_queue(&runtime->sleep, &d->wait);
- runtime->stop_threshold = d->stop_threshold;
+ if (!nonblock) {
+ for (i = 0; i < num_drecs; i++) {
+ d = &drec[i];
+ runtime = d->substream->runtime;
+ remove_wait_queue(&runtime->sleep, &d->wait);
+ }
+ if (drec != &drec_tmp)
+ kfree(drec);
}
-
- if (drec != &drec_tmp)
- kfree(drec);
snd_power_unlock(card);
return result;
@@ -2208,6 +2214,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst
case SNDRV_PCM_STATE_XRUN:
ret = -EPIPE;
goto __end;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ ret = -ESTRPIPE;
+ goto __end;
default:
ret = -EBADFD;
goto __end;
@@ -2253,6 +2262,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr
case SNDRV_PCM_STATE_XRUN:
ret = -EPIPE;
goto __end;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ ret = -ESTRPIPE;
+ goto __end;
default:
ret = -EBADFD;
goto __end;
@@ -2299,6 +2311,9 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs
case SNDRV_PCM_STATE_XRUN:
ret = -EPIPE;
goto __end;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ ret = -ESTRPIPE;
+ goto __end;
default:
ret = -EBADFD;
goto __end;
@@ -2345,6 +2360,9 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst
case SNDRV_PCM_STATE_XRUN:
ret = -EPIPE;
goto __end;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ ret = -ESTRPIPE;
+ goto __end;
default:
ret = -EBADFD;
goto __end;
@@ -2544,7 +2562,7 @@ static int snd_pcm_common_ioctl1(struct file *file,
return snd_pcm_hw_params_old_user(substream, arg);
#endif
case SNDRV_PCM_IOCTL_DRAIN:
- return snd_pcm_drain(substream);
+ return snd_pcm_drain(substream, file);
case SNDRV_PCM_IOCTL_DROP:
return snd_pcm_drop(substream);
case SNDRV_PCM_IOCTL_PAUSE:
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 473247c..c0adc14 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -274,7 +274,7 @@ static int open_substream(struct snd_rawmidi *rmidi,
return err;
substream->opened = 1;
if (substream->use_count++ == 0)
- substream->active_sensing = 1;
+ substream->active_sensing = 0;
if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
substream->append = 1;
rmidi->streams[substream->stream].substream_opened++;
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index 0a711d2..9dfb2f7 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -20,6 +20,7 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
+#include <sound/asoundef.h>
#include "seq_oss_midi.h"
#include "seq_oss_readq.h"
#include "seq_oss_timer.h"
@@ -476,19 +477,20 @@ snd_seq_oss_midi_reset(struct seq_oss_devinfo *dp, int dev)
ev.source.port = dp->port;
if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_SYNTH) {
ev.type = SNDRV_SEQ_EVENT_SENSING;
- snd_seq_oss_dispatch(dp, &ev, 0, 0); /* active sensing */
+ snd_seq_oss_dispatch(dp, &ev, 0, 0);
}
for (c = 0; c < 16; c++) {
ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
ev.data.control.channel = c;
- ev.data.control.param = 123;
- snd_seq_oss_dispatch(dp, &ev, 0, 0); /* all notes off */
+ ev.data.control.param = MIDI_CTL_ALL_NOTES_OFF;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0);
if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC) {
- ev.data.control.param = 121;
- snd_seq_oss_dispatch(dp, &ev, 0, 0); /* reset all controllers */
+ ev.data.control.param =
+ MIDI_CTL_RESET_CONTROLLERS;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0);
ev.type = SNDRV_SEQ_EVENT_PITCHBEND;
ev.data.control.value = 0;
- snd_seq_oss_dispatch(dp, &ev, 0, 0); /* bender off */
+ snd_seq_oss_dispatch(dp, &ev, 0, 0);
}
}
}
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
index 4d26146..ebaf1b5 100644
--- a/sound/core/seq/seq_midi.c
+++ b/sound/core/seq/seq_midi.c
@@ -120,7 +120,8 @@ static int dump_midi(struct snd_rawmidi_substream *substream, const char *buf, i
return -EINVAL;
runtime = substream->runtime;
if ((tmp = runtime->avail) < count) {
- snd_printd("warning, output event was lost (count = %i, available = %i)\n", count, tmp);
+ if (printk_ratelimit())
+ snd_printk(KERN_ERR "MIDI output buffer overrun\n");
return -ENOMEM;
}
if (snd_rawmidi_kernel_write(substream, buf, count) < count)
@@ -236,6 +237,7 @@ static int midisynth_use(void *private_data, struct snd_seq_port_subscribe *info
memset(&params, 0, sizeof(params));
params.avail_min = 1;
params.buffer_size = output_buffer_size;
+ params.no_active_sensing = 1;
if ((err = snd_rawmidi_output_params(msynth->output_rfile.output, &params)) < 0) {
snd_rawmidi_kernel_release(&msynth->output_rfile);
return err;
@@ -248,12 +250,9 @@ static int midisynth_use(void *private_data, struct snd_seq_port_subscribe *info
static int midisynth_unuse(void *private_data, struct snd_seq_port_subscribe *info)
{
struct seq_midisynth *msynth = private_data;
- unsigned char buf = 0xff; /* MIDI reset */
if (snd_BUG_ON(!msynth->output_rfile.output))
return -EINVAL;
- /* sending single MIDI reset message to shut the device up */
- snd_rawmidi_kernel_write(msynth->output_rfile.output, &buf, 1);
snd_rawmidi_drain_output(msynth->output_rfile.output);
return snd_rawmidi_kernel_release(&msynth->output_rfile);
}
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 257624b..3b9b550 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -353,7 +353,8 @@ static void master_free(struct snd_kcontrol *kcontrol)
*
* The optional argument @tlv can be used to specify the TLV information
* for dB scale of the master control. It should be a single element
- * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB.
+ * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or
+ * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB.
*/
struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
const unsigned int *tlv)
@@ -384,7 +385,10 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
kctl->private_free = master_free;
/* additional (constant) TLV read */
- if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) {
+ if (tlv &&
+ (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE ||
+ tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX ||
+ tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) {
kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
memcpy(master->tlv, tlv, sizeof(master->tlv));
kctl->tlv.p = master->tlv;
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 54239d2..6ba066c 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -25,12 +25,15 @@
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/wait.h>
+#include <linux/hrtimer.h>
+#include <linux/math64.h>
#include <linux/moduleparam.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/tlv.h>
#include <sound/pcm.h>
#include <sound/rawmidi.h>
+#include <sound/info.h>
#include <sound/initval.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
@@ -39,7 +42,7 @@ MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}");
#define MAX_PCM_DEVICES 4
-#define MAX_PCM_SUBSTREAMS 16
+#define MAX_PCM_SUBSTREAMS 128
#define MAX_MIDI_DEVICES 2
#if 0 /* emu10k1 emulation */
@@ -148,6 +151,10 @@ static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0};
static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8};
//static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+#ifdef CONFIG_HIGH_RES_TIMERS
+static int hrtimer = 1;
+#endif
+static int fake_buffer = 1;
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for dummy soundcard.");
@@ -161,6 +168,12 @@ module_param_array(pcm_substreams, int, NULL, 0444);
MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver.");
//module_param_array(midi_devs, int, NULL, 0444);
//MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver.");
+module_param(fake_buffer, bool, 0444);
+MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations.");
+#ifdef CONFIG_HIGH_RES_TIMERS
+module_param(hrtimer, bool, 0644);
+MODULE_PARM_DESC(hrtimer, "Use hrtimer as the timer source.");
+#endif
static struct platform_device *devices[SNDRV_CARDS];
@@ -171,137 +184,324 @@ static struct platform_device *devices[SNDRV_CARDS];
#define MIXER_ADDR_CD 4
#define MIXER_ADDR_LAST 4
+struct dummy_timer_ops {
+ int (*create)(struct snd_pcm_substream *);
+ void (*free)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+ int (*start)(struct snd_pcm_substream *);
+ int (*stop)(struct snd_pcm_substream *);
+ snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *);
+};
+
struct snd_dummy {
struct snd_card *card;
struct snd_pcm *pcm;
spinlock_t mixer_lock;
int mixer_volume[MIXER_ADDR_LAST+1][2];
int capture_source[MIXER_ADDR_LAST+1][2];
+ const struct dummy_timer_ops *timer_ops;
};
-struct snd_dummy_pcm {
- struct snd_dummy *dummy;
+/*
+ * system timer interface
+ */
+
+struct dummy_systimer_pcm {
spinlock_t lock;
struct timer_list timer;
- unsigned int pcm_buffer_size;
- unsigned int pcm_period_size;
- unsigned int pcm_bps; /* bytes per second */
- unsigned int pcm_hz; /* HZ */
- unsigned int pcm_irq_pos; /* IRQ position */
- unsigned int pcm_buf_pos; /* position in buffer */
+ unsigned long base_time;
+ unsigned int frac_pos; /* fractional sample position (based HZ) */
+ unsigned int frac_period_rest;
+ unsigned int frac_buffer_size; /* buffer_size * HZ */
+ unsigned int frac_period_size; /* period_size * HZ */
+ unsigned int rate;
+ int elapsed;
struct snd_pcm_substream *substream;
};
-
-static inline void snd_card_dummy_pcm_timer_start(struct snd_dummy_pcm *dpcm)
+static void dummy_systimer_rearm(struct dummy_systimer_pcm *dpcm)
{
- dpcm->timer.expires = 1 + jiffies;
+ dpcm->timer.expires = jiffies +
+ (dpcm->frac_period_rest + dpcm->rate - 1) / dpcm->rate;
add_timer(&dpcm->timer);
}
-static inline void snd_card_dummy_pcm_timer_stop(struct snd_dummy_pcm *dpcm)
+static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm)
{
- del_timer(&dpcm->timer);
+ unsigned long delta;
+
+ delta = jiffies - dpcm->base_time;
+ if (!delta)
+ return;
+ dpcm->base_time += delta;
+ delta *= dpcm->rate;
+ dpcm->frac_pos += delta;
+ while (dpcm->frac_pos >= dpcm->frac_buffer_size)
+ dpcm->frac_pos -= dpcm->frac_buffer_size;
+ while (dpcm->frac_period_rest <= delta) {
+ dpcm->elapsed++;
+ dpcm->frac_period_rest += dpcm->frac_period_size;
+ }
+ dpcm->frac_period_rest -= delta;
}
-static int snd_card_dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+static int dummy_systimer_start(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_dummy_pcm *dpcm = runtime->private_data;
- int err = 0;
+ struct dummy_systimer_pcm *dpcm = substream->runtime->private_data;
+ spin_lock(&dpcm->lock);
+ dpcm->base_time = jiffies;
+ dummy_systimer_rearm(dpcm);
+ spin_unlock(&dpcm->lock);
+ return 0;
+}
+static int dummy_systimer_stop(struct snd_pcm_substream *substream)
+{
+ struct dummy_systimer_pcm *dpcm = substream->runtime->private_data;
spin_lock(&dpcm->lock);
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- snd_card_dummy_pcm_timer_start(dpcm);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- snd_card_dummy_pcm_timer_stop(dpcm);
- break;
- default:
- err = -EINVAL;
- break;
- }
+ del_timer(&dpcm->timer);
spin_unlock(&dpcm->lock);
return 0;
}
-static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream)
+static int dummy_systimer_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_dummy_pcm *dpcm = runtime->private_data;
- int bps;
-
- bps = snd_pcm_format_width(runtime->format) * runtime->rate *
- runtime->channels / 8;
-
- if (bps <= 0)
- return -EINVAL;
-
- dpcm->pcm_bps = bps;
- dpcm->pcm_hz = HZ;
- dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream);
- dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream);
- dpcm->pcm_irq_pos = 0;
- dpcm->pcm_buf_pos = 0;
+ struct dummy_systimer_pcm *dpcm = runtime->private_data;
- snd_pcm_format_set_silence(runtime->format, runtime->dma_area,
- bytes_to_samples(runtime, runtime->dma_bytes));
+ dpcm->frac_pos = 0;
+ dpcm->rate = runtime->rate;
+ dpcm->frac_buffer_size = runtime->buffer_size * HZ;
+ dpcm->frac_period_size = runtime->period_size * HZ;
+ dpcm->frac_period_rest = dpcm->frac_period_size;
+ dpcm->elapsed = 0;
return 0;
}
-static void snd_card_dummy_pcm_timer_function(unsigned long data)
+static void dummy_systimer_callback(unsigned long data)
{
- struct snd_dummy_pcm *dpcm = (struct snd_dummy_pcm *)data;
+ struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data;
unsigned long flags;
+ int elapsed = 0;
spin_lock_irqsave(&dpcm->lock, flags);
- dpcm->timer.expires = 1 + jiffies;
- add_timer(&dpcm->timer);
- dpcm->pcm_irq_pos += dpcm->pcm_bps;
- dpcm->pcm_buf_pos += dpcm->pcm_bps;
- dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size * dpcm->pcm_hz;
- if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) {
- dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz;
- spin_unlock_irqrestore(&dpcm->lock, flags);
+ dummy_systimer_update(dpcm);
+ dummy_systimer_rearm(dpcm);
+ elapsed = dpcm->elapsed;
+ dpcm->elapsed = 0;
+ spin_unlock_irqrestore(&dpcm->lock, flags);
+ if (elapsed)
+ snd_pcm_period_elapsed(dpcm->substream);
+}
+
+static snd_pcm_uframes_t
+dummy_systimer_pointer(struct snd_pcm_substream *substream)
+{
+ struct dummy_systimer_pcm *dpcm = substream->runtime->private_data;
+ snd_pcm_uframes_t pos;
+
+ spin_lock(&dpcm->lock);
+ dummy_systimer_update(dpcm);
+ pos = dpcm->frac_pos / HZ;
+ spin_unlock(&dpcm->lock);
+ return pos;
+}
+
+static int dummy_systimer_create(struct snd_pcm_substream *substream)
+{
+ struct dummy_systimer_pcm *dpcm;
+
+ dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL);
+ if (!dpcm)
+ return -ENOMEM;
+ substream->runtime->private_data = dpcm;
+ init_timer(&dpcm->timer);
+ dpcm->timer.data = (unsigned long) dpcm;
+ dpcm->timer.function = dummy_systimer_callback;
+ spin_lock_init(&dpcm->lock);
+ dpcm->substream = substream;
+ return 0;
+}
+
+static void dummy_systimer_free(struct snd_pcm_substream *substream)
+{
+ kfree(substream->runtime->private_data);
+}
+
+static struct dummy_timer_ops dummy_systimer_ops = {
+ .create = dummy_systimer_create,
+ .free = dummy_systimer_free,
+ .prepare = dummy_systimer_prepare,
+ .start = dummy_systimer_start,
+ .stop = dummy_systimer_stop,
+ .pointer = dummy_systimer_pointer,
+};
+
+#ifdef CONFIG_HIGH_RES_TIMERS
+/*
+ * hrtimer interface
+ */
+
+struct dummy_hrtimer_pcm {
+ ktime_t base_time;
+ ktime_t period_time;
+ atomic_t running;
+ struct hrtimer timer;
+ struct tasklet_struct tasklet;
+ struct snd_pcm_substream *substream;
+};
+
+static void dummy_hrtimer_pcm_elapsed(unsigned long priv)
+{
+ struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv;
+ if (atomic_read(&dpcm->running))
snd_pcm_period_elapsed(dpcm->substream);
- } else
- spin_unlock_irqrestore(&dpcm->lock, flags);
}
-static snd_pcm_uframes_t snd_card_dummy_pcm_pointer(struct snd_pcm_substream *substream)
+static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer)
+{
+ struct dummy_hrtimer_pcm *dpcm;
+
+ dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer);
+ if (!atomic_read(&dpcm->running))
+ return HRTIMER_NORESTART;
+ tasklet_schedule(&dpcm->tasklet);
+ hrtimer_forward_now(timer, dpcm->period_time);
+ return HRTIMER_RESTART;
+}
+
+static int dummy_hrtimer_start(struct snd_pcm_substream *substream)
+{
+ struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data;
+
+ dpcm->base_time = hrtimer_cb_get_time(&dpcm->timer);
+ hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL);
+ atomic_set(&dpcm->running, 1);
+ return 0;
+}
+
+static int dummy_hrtimer_stop(struct snd_pcm_substream *substream)
+{
+ struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data;
+
+ atomic_set(&dpcm->running, 0);
+ hrtimer_cancel(&dpcm->timer);
+ return 0;
+}
+
+static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm)
+{
+ tasklet_kill(&dpcm->tasklet);
+}
+
+static snd_pcm_uframes_t
+dummy_hrtimer_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_dummy_pcm *dpcm = runtime->private_data;
+ struct dummy_hrtimer_pcm *dpcm = runtime->private_data;
+ u64 delta;
+ u32 pos;
+
+ delta = ktime_us_delta(hrtimer_cb_get_time(&dpcm->timer),
+ dpcm->base_time);
+ delta = div_u64(delta * runtime->rate + 999999, 1000000);
+ div_u64_rem(delta, runtime->buffer_size, &pos);
+ return pos;
+}
- return bytes_to_frames(runtime, dpcm->pcm_buf_pos / dpcm->pcm_hz);
+static int dummy_hrtimer_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct dummy_hrtimer_pcm *dpcm = runtime->private_data;
+ unsigned int period, rate;
+ long sec;
+ unsigned long nsecs;
+
+ dummy_hrtimer_sync(dpcm);
+ period = runtime->period_size;
+ rate = runtime->rate;
+ sec = period / rate;
+ period %= rate;
+ nsecs = div_u64((u64)period * 1000000000UL + rate - 1, rate);
+ dpcm->period_time = ktime_set(sec, nsecs);
+
+ return 0;
}
-static struct snd_pcm_hardware snd_card_dummy_playback =
+static int dummy_hrtimer_create(struct snd_pcm_substream *substream)
{
- .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
- .formats = USE_FORMATS,
- .rates = USE_RATE,
- .rate_min = USE_RATE_MIN,
- .rate_max = USE_RATE_MAX,
- .channels_min = USE_CHANNELS_MIN,
- .channels_max = USE_CHANNELS_MAX,
- .buffer_bytes_max = MAX_BUFFER_SIZE,
- .period_bytes_min = 64,
- .period_bytes_max = MAX_PERIOD_SIZE,
- .periods_min = USE_PERIODS_MIN,
- .periods_max = USE_PERIODS_MAX,
- .fifo_size = 0,
+ struct dummy_hrtimer_pcm *dpcm;
+
+ dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL);
+ if (!dpcm)
+ return -ENOMEM;
+ substream->runtime->private_data = dpcm;
+ hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ dpcm->timer.function = dummy_hrtimer_callback;
+ dpcm->substream = substream;
+ atomic_set(&dpcm->running, 0);
+ tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed,
+ (unsigned long)dpcm);
+ return 0;
+}
+
+static void dummy_hrtimer_free(struct snd_pcm_substream *substream)
+{
+ struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data;
+ dummy_hrtimer_sync(dpcm);
+ kfree(dpcm);
+}
+
+static struct dummy_timer_ops dummy_hrtimer_ops = {
+ .create = dummy_hrtimer_create,
+ .free = dummy_hrtimer_free,
+ .prepare = dummy_hrtimer_prepare,
+ .start = dummy_hrtimer_start,
+ .stop = dummy_hrtimer_stop,
+ .pointer = dummy_hrtimer_pointer,
};
-static struct snd_pcm_hardware snd_card_dummy_capture =
+#endif /* CONFIG_HIGH_RES_TIMERS */
+
+/*
+ * PCM interface
+ */
+
+static int dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
- .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ struct snd_dummy *dummy = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ return dummy->timer_ops->start(substream);
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ return dummy->timer_ops->stop(substream);
+ }
+ return -EINVAL;
+}
+
+static int dummy_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dummy *dummy = snd_pcm_substream_chip(substream);
+
+ return dummy->timer_ops->prepare(substream);
+}
+
+static snd_pcm_uframes_t dummy_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_dummy *dummy = snd_pcm_substream_chip(substream);
+
+ return dummy->timer_ops->pointer(substream);
+}
+
+static struct snd_pcm_hardware dummy_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
.formats = USE_FORMATS,
.rates = USE_RATE,
.rate_min = USE_RATE_MIN,
@@ -316,123 +516,152 @@ static struct snd_pcm_hardware snd_card_dummy_capture =
.fifo_size = 0,
};
-static void snd_card_dummy_runtime_free(struct snd_pcm_runtime *runtime)
-{
- kfree(runtime->private_data);
-}
-
-static int snd_card_dummy_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *hw_params)
+static int dummy_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
{
- return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (fake_buffer) {
+ /* runtime->dma_bytes has to be set manually to allow mmap */
+ substream->runtime->dma_bytes = params_buffer_bytes(hw_params);
+ return 0;
+ }
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
}
-static int snd_card_dummy_hw_free(struct snd_pcm_substream *substream)
+static int dummy_pcm_hw_free(struct snd_pcm_substream *substream)
{
+ if (fake_buffer)
+ return 0;
return snd_pcm_lib_free_pages(substream);
}
-static struct snd_dummy_pcm *new_pcm_stream(struct snd_pcm_substream *substream)
-{
- struct snd_dummy_pcm *dpcm;
-
- dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL);
- if (! dpcm)
- return dpcm;
- init_timer(&dpcm->timer);
- dpcm->timer.data = (unsigned long) dpcm;
- dpcm->timer.function = snd_card_dummy_pcm_timer_function;
- spin_lock_init(&dpcm->lock);
- dpcm->substream = substream;
- return dpcm;
-}
-
-static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream)
+static int dummy_pcm_open(struct snd_pcm_substream *substream)
{
+ struct snd_dummy *dummy = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_dummy_pcm *dpcm;
int err;
- if ((dpcm = new_pcm_stream(substream)) == NULL)
- return -ENOMEM;
- runtime->private_data = dpcm;
- /* makes the infrastructure responsible for freeing dpcm */
- runtime->private_free = snd_card_dummy_runtime_free;
- runtime->hw = snd_card_dummy_playback;
+ dummy->timer_ops = &dummy_systimer_ops;
+#ifdef CONFIG_HIGH_RES_TIMERS
+ if (hrtimer)
+ dummy->timer_ops = &dummy_hrtimer_ops;
+#endif
+
+ err = dummy->timer_ops->create(substream);
+ if (err < 0)
+ return err;
+
+ runtime->hw = dummy_pcm_hardware;
if (substream->pcm->device & 1) {
runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
}
if (substream->pcm->device & 2)
- runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID);
- err = add_playback_constraints(runtime);
- if (err < 0)
+ runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ err = add_playback_constraints(substream->runtime);
+ else
+ err = add_capture_constraints(substream->runtime);
+ if (err < 0) {
+ dummy->timer_ops->free(substream);
return err;
-
+ }
return 0;
}
-static int snd_card_dummy_capture_open(struct snd_pcm_substream *substream)
+static int dummy_pcm_close(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_dummy_pcm *dpcm;
- int err;
+ struct snd_dummy *dummy = snd_pcm_substream_chip(substream);
+ dummy->timer_ops->free(substream);
+ return 0;
+}
- if ((dpcm = new_pcm_stream(substream)) == NULL)
- return -ENOMEM;
- runtime->private_data = dpcm;
- /* makes the infrastructure responsible for freeing dpcm */
- runtime->private_free = snd_card_dummy_runtime_free;
- runtime->hw = snd_card_dummy_capture;
- if (substream->pcm->device == 1) {
- runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
- runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
+/*
+ * dummy buffer handling
+ */
+
+static void *dummy_page[2];
+
+static void free_fake_buffer(void)
+{
+ if (fake_buffer) {
+ int i;
+ for (i = 0; i < 2; i++)
+ if (dummy_page[i]) {
+ free_page((unsigned long)dummy_page[i]);
+ dummy_page[i] = NULL;
+ }
}
- if (substream->pcm->device & 2)
- runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID);
- err = add_capture_constraints(runtime);
- if (err < 0)
- return err;
+}
+static int alloc_fake_buffer(void)
+{
+ int i;
+
+ if (!fake_buffer)
+ return 0;
+ for (i = 0; i < 2; i++) {
+ dummy_page[i] = (void *)get_zeroed_page(GFP_KERNEL);
+ if (!dummy_page[i]) {
+ free_fake_buffer();
+ return -ENOMEM;
+ }
+ }
return 0;
}
-static int snd_card_dummy_playback_close(struct snd_pcm_substream *substream)
+static int dummy_pcm_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ void __user *dst, snd_pcm_uframes_t count)
{
- return 0;
+ return 0; /* do nothing */
}
-static int snd_card_dummy_capture_close(struct snd_pcm_substream *substream)
+static int dummy_pcm_silence(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
{
- return 0;
+ return 0; /* do nothing */
+}
+
+static struct page *dummy_pcm_page(struct snd_pcm_substream *substream,
+ unsigned long offset)
+{
+ return virt_to_page(dummy_page[substream->stream]); /* the same page */
}
-static struct snd_pcm_ops snd_card_dummy_playback_ops = {
- .open = snd_card_dummy_playback_open,
- .close = snd_card_dummy_playback_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_card_dummy_hw_params,
- .hw_free = snd_card_dummy_hw_free,
- .prepare = snd_card_dummy_pcm_prepare,
- .trigger = snd_card_dummy_pcm_trigger,
- .pointer = snd_card_dummy_pcm_pointer,
+static struct snd_pcm_ops dummy_pcm_ops = {
+ .open = dummy_pcm_open,
+ .close = dummy_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = dummy_pcm_hw_params,
+ .hw_free = dummy_pcm_hw_free,
+ .prepare = dummy_pcm_prepare,
+ .trigger = dummy_pcm_trigger,
+ .pointer = dummy_pcm_pointer,
};
-static struct snd_pcm_ops snd_card_dummy_capture_ops = {
- .open = snd_card_dummy_capture_open,
- .close = snd_card_dummy_capture_close,
- .ioctl = snd_pcm_lib_ioctl,
- .hw_params = snd_card_dummy_hw_params,
- .hw_free = snd_card_dummy_hw_free,
- .prepare = snd_card_dummy_pcm_prepare,
- .trigger = snd_card_dummy_pcm_trigger,
- .pointer = snd_card_dummy_pcm_pointer,
+static struct snd_pcm_ops dummy_pcm_ops_no_buf = {
+ .open = dummy_pcm_open,
+ .close = dummy_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = dummy_pcm_hw_params,
+ .hw_free = dummy_pcm_hw_free,
+ .prepare = dummy_pcm_prepare,
+ .trigger = dummy_pcm_trigger,
+ .pointer = dummy_pcm_pointer,
+ .copy = dummy_pcm_copy,
+ .silence = dummy_pcm_silence,
+ .page = dummy_pcm_page,
};
static int __devinit snd_card_dummy_pcm(struct snd_dummy *dummy, int device,
int substreams)
{
struct snd_pcm *pcm;
+ struct snd_pcm_ops *ops;
int err;
err = snd_pcm_new(dummy->card, "Dummy PCM", device,
@@ -440,17 +669,28 @@ static int __devinit snd_card_dummy_pcm(struct snd_dummy *dummy, int device,
if (err < 0)
return err;
dummy->pcm = pcm;
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_dummy_playback_ops);
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_dummy_capture_ops);
+ if (fake_buffer)
+ ops = &dummy_pcm_ops_no_buf;
+ else
+ ops = &dummy_pcm_ops;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, ops);
pcm->private_data = dummy;
pcm->info_flags = 0;
strcpy(pcm->name, "Dummy PCM");
- snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
- snd_dma_continuous_data(GFP_KERNEL),
- 0, 64*1024);
+ if (!fake_buffer) {
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, 64*1024);
+ }
return 0;
}
+/*
+ * mixer interface
+ */
+
#define DUMMY_VOLUME(xname, xindex, addr) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
@@ -581,6 +821,131 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy)
return 0;
}
+#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_PROC_FS)
+/*
+ * proc interface
+ */
+static void print_formats(struct snd_info_buffer *buffer)
+{
+ int i;
+
+ for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
+ if (dummy_pcm_hardware.formats & (1ULL << i))
+ snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
+ }
+}
+
+static void print_rates(struct snd_info_buffer *buffer)
+{
+ static int rates[] = {
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000,
+ 64000, 88200, 96000, 176400, 192000,
+ };
+ int i;
+
+ if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_CONTINUOUS)
+ snd_iprintf(buffer, " continuous");
+ if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_KNOT)
+ snd_iprintf(buffer, " knot");
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
+ if (dummy_pcm_hardware.rates & (1 << i))
+ snd_iprintf(buffer, " %d", rates[i]);
+}
+
+#define get_dummy_int_ptr(ofs) \
+ (unsigned int *)((char *)&dummy_pcm_hardware + (ofs))
+#define get_dummy_ll_ptr(ofs) \
+ (unsigned long long *)((char *)&dummy_pcm_hardware + (ofs))
+
+struct dummy_hw_field {
+ const char *name;
+ const char *format;
+ unsigned int offset;
+ unsigned int size;
+};
+#define FIELD_ENTRY(item, fmt) { \
+ .name = #item, \
+ .format = fmt, \
+ .offset = offsetof(struct snd_pcm_hardware, item), \
+ .size = sizeof(dummy_pcm_hardware.item) }
+
+static struct dummy_hw_field fields[] = {
+ FIELD_ENTRY(formats, "%#llx"),
+ FIELD_ENTRY(rates, "%#x"),
+ FIELD_ENTRY(rate_min, "%d"),
+ FIELD_ENTRY(rate_max, "%d"),
+ FIELD_ENTRY(channels_min, "%d"),
+ FIELD_ENTRY(channels_max, "%d"),
+ FIELD_ENTRY(buffer_bytes_max, "%ld"),
+ FIELD_ENTRY(period_bytes_min, "%ld"),
+ FIELD_ENTRY(period_bytes_max, "%ld"),
+ FIELD_ENTRY(periods_min, "%d"),
+ FIELD_ENTRY(periods_max, "%d"),
+};
+
+static void dummy_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(fields); i++) {
+ snd_iprintf(buffer, "%s ", fields[i].name);
+ if (fields[i].size == sizeof(int))
+ snd_iprintf(buffer, fields[i].format,
+ *get_dummy_int_ptr(fields[i].offset));
+ else
+ snd_iprintf(buffer, fields[i].format,
+ *get_dummy_ll_ptr(fields[i].offset));
+ if (!strcmp(fields[i].name, "formats"))
+ print_formats(buffer);
+ else if (!strcmp(fields[i].name, "rates"))
+ print_rates(buffer);
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void dummy_proc_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ char line[64];
+
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ char item[20];
+ const char *ptr;
+ unsigned long long val;
+ int i;
+
+ ptr = snd_info_get_str(item, line, sizeof(item));
+ for (i = 0; i < ARRAY_SIZE(fields); i++) {
+ if (!strcmp(item, fields[i].name))
+ break;
+ }
+ if (i >= ARRAY_SIZE(fields))
+ continue;
+ snd_info_get_str(item, ptr, sizeof(item));
+ if (strict_strtoull(item, 0, &val))
+ continue;
+ if (fields[i].size == sizeof(int))
+ *get_dummy_int_ptr(fields[i].offset) = val;
+ else
+ *get_dummy_ll_ptr(fields[i].offset) = val;
+ }
+}
+
+static void __devinit dummy_proc_init(struct snd_dummy *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(chip->card, "dummy_pcm", &entry)) {
+ snd_info_set_text_ops(entry, chip, dummy_proc_read);
+ entry->c.text.write = dummy_proc_write;
+ entry->mode |= S_IWUSR;
+ }
+}
+#else
+#define dummy_proc_init(x)
+#endif /* CONFIG_SND_DEBUG && CONFIG_PROC_FS */
+
static int __devinit snd_dummy_probe(struct platform_device *devptr)
{
struct snd_card *card;
@@ -610,6 +975,8 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr)
strcpy(card->shortname, "Dummy");
sprintf(card->longname, "Dummy %i", dev + 1);
+ dummy_proc_init(dummy);
+
snd_card_set_dev(card, &devptr->dev);
err = snd_card_register(card);
@@ -670,6 +1037,7 @@ static void snd_dummy_unregister_all(void)
for (i = 0; i < ARRAY_SIZE(devices); ++i)
platform_device_unregister(devices[i]);
platform_driver_unregister(&snd_dummy_driver);
+ free_fake_buffer();
}
static int __init alsa_card_dummy_init(void)
@@ -680,6 +1048,12 @@ static int __init alsa_card_dummy_init(void)
if (err < 0)
return err;
+ err = alloc_fake_buffer();
+ if (err < 0) {
+ platform_driver_unregister(&snd_dummy_driver);
+ return err;
+ }
+
cards = 0;
for (i = 0; i < SNDRV_CARDS; i++) {
struct platform_device *device;
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index 3ee0269..02f79d2 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -1,5 +1,5 @@
/*
- * Driver for C-Media's CMI8330 soundcards.
+ * Driver for C-Media's CMI8330 and CMI8329 soundcards.
* Copyright (c) by George Talusan <gstalusan@uwaterloo.ca>
* http://www.undergrad.math.uwaterloo.ca/~gstalusa
*
@@ -35,7 +35,7 @@
*
* This card has two mixers and two PCM devices. I've cheesed it such
* that recording and playback can be done through the same device.
- * The driver "magically" routes the capturing to the CMI8330 codec,
+ * The driver "magically" routes the capturing to the AD1848 codec,
* and playback to the SB16 codec. This allows for full-duplex mode
* to some extent.
* The utilities in alsa-utils are aware of both devices, so passing
@@ -64,7 +64,7 @@
/*
*/
MODULE_AUTHOR("George Talusan <gstalusan@uwaterloo.ca>");
-MODULE_DESCRIPTION("C-Media CMI8330");
+MODULE_DESCRIPTION("C-Media CMI8330/CMI8329");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}");
@@ -86,38 +86,38 @@ static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
module_param_array(index, int, NULL, 0444);
-MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard.");
+MODULE_PARM_DESC(index, "Index value for CMI8330/CMI8329 soundcard.");
module_param_array(id, charp, NULL, 0444);
-MODULE_PARM_DESC(id, "ID string for CMI8330 soundcard.");
+MODULE_PARM_DESC(id, "ID string for CMI8330/CMI8329 soundcard.");
module_param_array(enable, bool, NULL, 0444);
-MODULE_PARM_DESC(enable, "Enable CMI8330 soundcard.");
+MODULE_PARM_DESC(enable, "Enable CMI8330/CMI8329 soundcard.");
#ifdef CONFIG_PNP
module_param_array(isapnp, bool, NULL, 0444);
MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard.");
#endif
module_param_array(sbport, long, NULL, 0444);
-MODULE_PARM_DESC(sbport, "Port # for CMI8330 SB driver.");
+MODULE_PARM_DESC(sbport, "Port # for CMI8330/CMI8329 SB driver.");
module_param_array(sbirq, int, NULL, 0444);
-MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330 SB driver.");
+MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330/CMI8329 SB driver.");
module_param_array(sbdma8, int, NULL, 0444);
-MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330 SB driver.");
+MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330/CMI8329 SB driver.");
module_param_array(sbdma16, int, NULL, 0444);
-MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330 SB driver.");
+MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330/CMI8329 SB driver.");
module_param_array(wssport, long, NULL, 0444);
-MODULE_PARM_DESC(wssport, "Port # for CMI8330 WSS driver.");
+MODULE_PARM_DESC(wssport, "Port # for CMI8330/CMI8329 WSS driver.");
module_param_array(wssirq, int, NULL, 0444);
-MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver.");
+MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330/CMI8329 WSS driver.");
module_param_array(wssdma, int, NULL, 0444);
-MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver.");
+MODULE_PARM_DESC(wssdma, "DMA for CMI8330/CMI8329 WSS driver.");
module_param_array(fmport, long, NULL, 0444);
-MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver.");
+MODULE_PARM_DESC(fmport, "FM port # for CMI8330/CMI8329 driver.");
module_param_array(mpuport, long, NULL, 0444);
-MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver.");
+MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330/CMI8329 driver.");
module_param_array(mpuirq, int, NULL, 0444);
-MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port.");
+MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330/CMI8329 MPU-401 port.");
#ifdef CONFIG_PNP
static int isa_registered;
static int pnp_registered;
@@ -156,6 +156,11 @@ static unsigned char snd_cmi8330_image[((CMI8330_CDINGAIN)-16) + 1] =
typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *);
+enum card_type {
+ CMI8330,
+ CMI8329
+};
+
struct snd_cmi8330 {
#ifdef CONFIG_PNP
struct pnp_dev *cap;
@@ -172,11 +177,14 @@ struct snd_cmi8330 {
snd_pcm_open_callback_t open;
void *private_data; /* sb or wss */
} streams[2];
+
+ enum card_type type;
};
#ifdef CONFIG_PNP
static struct pnp_card_device_id snd_cmi8330_pnpids[] = {
+ { .id = "CMI0001", .devs = { { "@X@0001" }, { "@@@0001" }, { "@H@0001" }, { "A@@0001" } } },
{ .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } },
{ .id = "" }
};
@@ -304,7 +312,7 @@ static int __devinit snd_cmi8330_mixer(struct snd_card *card, struct snd_cmi8330
unsigned int idx;
int err;
- strcpy(card->mixername, "CMI8330/C3D");
+ strcpy(card->mixername, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D");
for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) {
err = snd_ctl_add(card,
@@ -329,6 +337,9 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard,
struct pnp_dev *pdev;
int err;
+ /* CMI8329 has a device with ID A@@0001, CMI8330 does not */
+ acard->type = (id->devs[3].id[0]) ? CMI8329 : CMI8330;
+
acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL);
if (acard->cap == NULL)
return -EBUSY;
@@ -345,38 +356,45 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard,
err = pnp_activate_dev(pdev);
if (err < 0) {
- snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n");
+ snd_printk(KERN_ERR "AD1848 PnP configure failure\n");
return -EBUSY;
}
wssport[dev] = pnp_port_start(pdev, 0);
wssdma[dev] = pnp_dma(pdev, 0);
wssirq[dev] = pnp_irq(pdev, 0);
- fmport[dev] = pnp_port_start(pdev, 1);
+ if (pnp_port_start(pdev, 1))
+ fmport[dev] = pnp_port_start(pdev, 1);
/* allocate SB16 resources */
pdev = acard->play;
err = pnp_activate_dev(pdev);
if (err < 0) {
- snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n");
+ snd_printk(KERN_ERR "SB16 PnP configure failure\n");
return -EBUSY;
}
sbport[dev] = pnp_port_start(pdev, 0);
sbdma8[dev] = pnp_dma(pdev, 0);
sbdma16[dev] = pnp_dma(pdev, 1);
sbirq[dev] = pnp_irq(pdev, 0);
+ /* On CMI8239, the OPL3 port might be present in SB16 PnP resources */
+ if (fmport[dev] == SNDRV_AUTO_PORT) {
+ if (pnp_port_start(pdev, 1))
+ fmport[dev] = pnp_port_start(pdev, 1);
+ else
+ fmport[dev] = 0x388; /* Or hardwired */
+ }
/* allocate MPU-401 resources */
pdev = acard->mpu;
err = pnp_activate_dev(pdev);
- if (err < 0) {
- snd_printk(KERN_ERR
- "CMI8330/C3D (MPU-401) PnP configure failure\n");
- return -EBUSY;
+ if (err < 0)
+ snd_printk(KERN_ERR "MPU-401 PnP configure failure: will be disabled\n");
+ else {
+ mpuport[dev] = pnp_port_start(pdev, 0);
+ mpuirq[dev] = pnp_irq(pdev, 0);
}
- mpuport[dev] = pnp_port_start(pdev, 0);
- mpuirq[dev] = pnp_irq(pdev, 0);
return 0;
}
#endif
@@ -430,9 +448,9 @@ static int __devinit snd_cmi8330_pcm(struct snd_card *card, struct snd_cmi8330 *
snd_cmi8330_capture_open
};
- if ((err = snd_pcm_new(card, "CMI8330", 0, 1, 1, &pcm)) < 0)
+ if ((err = snd_pcm_new(card, (chip->type == CMI8329) ? "CMI8329" : "CMI8330", 0, 1, 1, &pcm)) < 0)
return err;
- strcpy(pcm->name, "CMI8330");
+ strcpy(pcm->name, (chip->type == CMI8329) ? "CMI8329" : "CMI8330");
pcm->private_data = chip;
/* SB16 */
@@ -527,11 +545,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
wssdma[dev], -1,
WSS_HW_DETECT, 0, &acard->wss);
if (err < 0) {
- snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n");
+ snd_printk(KERN_ERR PFX "AD1848 device busy??\n");
return err;
}
if (acard->wss->hardware != WSS_HW_CMI8330) {
- snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n");
+ snd_printk(KERN_ERR PFX "AD1848 not found during probe\n");
return -ENODEV;
}
@@ -541,11 +559,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
sbdma8[dev],
sbdma16[dev],
SB_HW_AUTO, &acard->sb)) < 0) {
- snd_printk(KERN_ERR PFX "(SB16) device busy??\n");
+ snd_printk(KERN_ERR PFX "SB16 device busy??\n");
return err;
}
if (acard->sb->hardware != SB_HW_16) {
- snd_printk(KERN_ERR PFX "(SB16) not found during probe\n");
+ snd_printk(KERN_ERR PFX "SB16 not found during probe\n");
return err;
}
@@ -585,8 +603,8 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
mpuport[dev]);
}
- strcpy(card->driver, "CMI8330/C3D");
- strcpy(card->shortname, "C-Media CMI8330/C3D");
+ strcpy(card->driver, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D");
+ strcpy(card->shortname, (acard->type == CMI8329) ? "C-Media CMI8329" : "C-Media CMI8330/C3D");
sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
card->shortname,
acard->wss->port,
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index a40be0c..782b3b8 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -127,15 +127,16 @@ static void midi_poll(unsigned long dummy)
for (dev = 0; dev < num_midis; dev++)
if (midi_devs[dev] != NULL && midi_out_buf[dev] != NULL)
{
- int ok = 1;
-
- while (DATA_AVAIL(midi_out_buf[dev]) && ok)
+ while (DATA_AVAIL(midi_out_buf[dev]))
{
+ int ok;
int c = midi_out_buf[dev]->queue[midi_out_buf[dev]->head];
spin_unlock_irqrestore(&lock,flags);/* Give some time to others */
ok = midi_devs[dev]->outputc(dev, c);
spin_lock_irqsave(&lock, flags);
+ if (!ok)
+ break;
midi_out_buf[dev]->head = (midi_out_buf[dev]->head + 1) % MAX_QUEUE_SIZE;
midi_out_buf[dev]->len--;
}
diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
index 187f727..6713110 100644
--- a/sound/oss/vwsnd.c
+++ b/sound/oss/vwsnd.c
@@ -628,7 +628,7 @@ static void li_setup_dma(dma_chan_t *chan,
ASSERT(!(buffer_paddr & 0xFF));
chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8);
- chan->cfgval = (!LI_CCFG_LOCK |
+ chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) |
SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) |
desc->direction |
mode |
@@ -638,9 +638,9 @@ static void li_setup_dma(dma_chan_t *chan,
tmask = 13 - fragshift; /* See Lithium DMA Notes above. */
ASSERT(size >= 2 && size <= 7);
ASSERT(tmask >= 1 && tmask <= 7);
- chan->ctlval = (!LI_CCTL_RESET |
+ chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) |
SHIFT_FIELD(size, LI_CCTL_SIZE) |
- !LI_CCTL_DMA_ENABLE |
+ (chan->ctlval & ~LI_CCTL_DMA_ENABLE) |
SHIFT_FIELD(tmask, LI_CCTL_TMASK) |
SHIFT_FIELD(0, LI_CCTL_TPTR));
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 748f6b7..fb5ee3c 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -135,11 +135,11 @@ config SND_AW2
config SND_AZT3328
- tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)"
- depends on EXPERIMENTAL
+ tristate "Aztech AZF3328 / PCI168"
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_PCM
+ select SND_RAWMIDI
help
Say Y here to include support for Aztech AZF3328 (PCI168)
soundcards.
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 76d76c0..b458d20 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -478,45 +478,6 @@ static int snd_ali_reset_5451(struct snd_ali *codec)
return 0;
}
-#ifdef CODEC_RESET
-
-static int snd_ali_reset_codec(struct snd_ali *codec)
-{
- struct pci_dev *pci_dev;
- unsigned char bVal;
- unsigned int dwVal;
- unsigned short wCount, wReg;
-
- pci_dev = codec->pci_m1533;
-
- pci_read_config_dword(pci_dev, 0x7c, &dwVal);
- pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000);
- udelay(5000);
- pci_read_config_dword(pci_dev, 0x7c, &dwVal);
- pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff);
- udelay(5000);
-
- bVal = inb(ALI_REG(codec,ALI_SCTRL));
- bVal |= 0x02;
- outb(ALI_REG(codec,ALI_SCTRL),bVal);
- udelay(5000);
- bVal = inb(ALI_REG(codec,ALI_SCTRL));
- bVal &= 0xfd;
- outb(ALI_REG(codec,ALI_SCTRL),bVal);
- udelay(15000);
-
- wCount = 200;
- while (wCount--) {
- wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN);
- if ((wReg & 0x000f) == 0x000f)
- return 0;
- udelay(5000);
- }
- return -1;
-}
-
-#endif
-
/*
* ALI 5451 Controller
*/
@@ -561,22 +522,6 @@ static void snd_ali_disable_address_interrupt(struct snd_ali *codec)
outl(gc, ALI_REG(codec, ALI_GC_CIR));
}
-#if 0 /* not used */
-static void snd_ali_enable_voice_irq(struct snd_ali *codec,
- unsigned int channel)
-{
- unsigned int mask;
- struct snd_ali_channel_control *pchregs = &(codec->chregs);
-
- snd_ali_printk("enable_voice_irq channel=%d\n",channel);
-
- mask = 1 << (channel & 0x1f);
- pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten));
- pchregs->data.ainten |= mask;
- outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten));
-}
-#endif
-
static void snd_ali_disable_voice_irq(struct snd_ali *codec,
unsigned int channel)
{
@@ -677,16 +622,6 @@ static void snd_ali_free_channel_pcm(struct snd_ali *codec, int channel)
}
}
-#if 0 /* not used */
-static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel)
-{
- unsigned int mask = 1 << (channel & 0x1f);
-
- snd_ali_printk("start_voice: channel=%d\n",channel);
- outl(mask, ALI_REG(codec,codec->chregs.regs.start));
-}
-#endif
-
static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel)
{
unsigned int mask = 1 << (channel & 0x1f);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index f290bc5..8451a01 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1,6 +1,6 @@
/*
* azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168).
- * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de>
+ * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr <andi AT lisas.de>
*
* Framework borrowed from Bart Hartgers's als4000.c.
* Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801),
@@ -10,6 +10,13 @@
* PCI168 A/AP, sub ID 8000
* Please give me feedback in case you try my driver with one of these!!
*
+ * Keywords: Windows XP Vista 168nt4-125.zip 168win95-125.zip PCI 168 download
+ * (XP/Vista do not support this card at all but every Linux distribution
+ * has very good support out of the box;
+ * just to make sure that the right people hit this and get to know that,
+ * despite the high level of Internet ignorance - as usual :-P -
+ * about very good support for this card - on Linux!)
+ *
* GPL LICENSE
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -71,10 +78,11 @@
* - built-in General DirectX timer having a 20 bits counter
* with 1us resolution (see below!)
* - I2S serial output port for external DAC
+ * [FIXME: 3.3V or 5V level? maximum rate is 66.2kHz right?]
* - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI
* - supports hardware volume control
* - single chip low cost solution (128 pin QFP)
- * - supports programmable Sub-vendor and Sub-system ID
+ * - supports programmable Sub-vendor and Sub-system ID [24C02 SEEPROM chip]
* required for Microsoft's logo compliance (FIXME: where?)
* At least the Trident 4D Wave DX has one bit somewhere
* to enable writes to PCI subsystem VID registers, that should be it.
@@ -82,6 +90,7 @@
* some custom data starting at 0x80. What kind of config settings
* are located in our extended PCI space anyway??
* - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms
+ * [TDA1517P chip]
*
* Note that this driver now is actually *better* than the Windows driver,
* since it additionally supports the card's 1MHz DirectX timer - just try
@@ -146,10 +155,15 @@
* to read the Digital Enhanced Game Port. Not sure whether it is fixable.
*
* TODO
+ * - use PCI_VDEVICE
+ * - verify driver status on x86_64
+ * - test multi-card driver operation
+ * - (ab)use 1MHz DirectX timer as kernel clocksource
* - test MPU401 MIDI playback etc.
* - add more power micro-management (disable various units of the card
- * as long as they're unused). However this requires more I/O ports which I
- * haven't figured out yet and which thus might not even exist...
+ * as long as they're unused, to improve audio quality and save power).
+ * However this requires more I/O ports which I haven't figured out yet
+ * and which thus might not even exist...
* The standard suspend/resume functionality could probably make use of
* some improvement, too...
* - figure out what all unknown port bits are responsible for
@@ -185,25 +199,46 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
#define SUPPORT_GAMEPORT 1
#endif
+/* === Debug settings ===
+ Further diagnostic functionality than the settings below
+ does not need to be provided, since one can easily write a bash script
+ to dump the card's I/O ports (those listed in lspci -v -v):
+ function dump()
+ {
+ local descr=$1; local addr=$2; local count=$3
+
+ echo "${descr}: ${count} @ ${addr}:"
+ dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C
+ }
+ and then use something like
+ "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8",
+ "dump codec00 0xa800 32", "dump mixer 0xb800 64", "dump synth 0xbc00 8",
+ possibly within a "while true; do ... sleep 1; done" loop.
+ Tweaking ports could be done using
+ VALSTRING="`printf "%02x" $value`"
+ printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null
+*/
+
#define DEBUG_MISC 0
#define DEBUG_CALLS 0
#define DEBUG_MIXER 0
-#define DEBUG_PLAY_REC 0
+#define DEBUG_CODEC 0
#define DEBUG_IO 0
#define DEBUG_TIMER 0
#define DEBUG_GAME 0
+#define DEBUG_PM 0
#define MIXER_TESTING 0
#if DEBUG_MISC
-#define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbgmisc(format, args...) printk(KERN_DEBUG format, ##args)
#else
#define snd_azf3328_dbgmisc(format, args...)
#endif
#if DEBUG_CALLS
#define snd_azf3328_dbgcalls(format, args...) printk(format, ##args)
-#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
-#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
+#define snd_azf3328_dbgcallenter() printk(KERN_DEBUG "--> %s\n", __func__)
+#define snd_azf3328_dbgcallleave() printk(KERN_DEBUG "<-- %s\n", __func__)
#else
#define snd_azf3328_dbgcalls(format, args...)
#define snd_azf3328_dbgcallenter()
@@ -216,10 +251,10 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
#define snd_azf3328_dbgmixer(format, args...)
#endif
-#if DEBUG_PLAY_REC
-#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args)
+#if DEBUG_CODEC
+#define snd_azf3328_dbgcodec(format, args...) printk(KERN_DEBUG format, ##args)
#else
-#define snd_azf3328_dbgplay(format, args...)
+#define snd_azf3328_dbgcodec(format, args...)
#endif
#if DEBUG_MISC
@@ -234,6 +269,12 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
#define snd_azf3328_dbggame(format, args...)
#endif
+#if DEBUG_PM
+#define snd_azf3328_dbgpm(format, args...) printk(KERN_DEBUG format, ##args)
+#else
+#define snd_azf3328_dbgpm(format, args...)
+#endif
+
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for AZF3328 soundcard.");
@@ -250,22 +291,23 @@ static int seqtimer_scaling = 128;
module_param(seqtimer_scaling, int, 0444);
MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128.");
-struct snd_azf3328_audio_stream {
+struct snd_azf3328_codec_data {
+ unsigned long io_base;
struct snd_pcm_substream *substream;
- int enabled;
- int running;
- unsigned long portbase;
+ bool running;
+ const char *name;
};
-enum snd_azf3328_stream_index {
- AZF_PLAYBACK = 0,
- AZF_CAPTURE = 1,
+enum snd_azf3328_codec_type {
+ AZF_CODEC_PLAYBACK = 0,
+ AZF_CODEC_CAPTURE = 1,
+ AZF_CODEC_I2S_OUT = 2,
};
struct snd_azf3328 {
/* often-used fields towards beginning, then grouped */
- unsigned long codec_io; /* usually 0xb000, size 128 */
+ unsigned long ctrl_io; /* usually 0xb000, size 128 */
unsigned long game_io; /* usually 0xb400, size 8 */
unsigned long mpu_io; /* usually 0xb800, size 4 */
unsigned long opl3_io; /* usually 0xbc00, size 8 */
@@ -275,15 +317,17 @@ struct snd_azf3328 {
struct snd_timer *timer;
- struct snd_pcm *pcm;
- struct snd_azf3328_audio_stream audio_stream[2];
+ struct snd_pcm *pcm[3];
+
+ /* playback, recording and I2S out codecs */
+ struct snd_azf3328_codec_data codecs[3];
struct snd_card *card;
struct snd_rawmidi *rmidi;
#ifdef SUPPORT_GAMEPORT
struct gameport *gameport;
- int axes[4];
+ u16 axes[4];
#endif
struct pci_dev *pci;
@@ -293,16 +337,16 @@ struct snd_azf3328 {
* If we need to add more registers here, then we might try to fold this
* into some transparent combined shadow register handling with
* CONFIG_PM register storage below, but that's slightly difficult. */
- u16 shadow_reg_codec_6AH;
+ u16 shadow_reg_ctrl_6AH;
#ifdef CONFIG_PM
/* register value containers for power management
- * Note: not always full I/O range preserved (just like Win driver!) */
- u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2];
- u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2];
- u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2];
- u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2];
- u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2];
+ * Note: not always full I/O range preserved (similar to Win driver!) */
+ u32 saved_regs_ctrl[AZF_ALIGN(AZF_IO_SIZE_CTRL_PM) / 4];
+ u32 saved_regs_game[AZF_ALIGN(AZF_IO_SIZE_GAME_PM) / 4];
+ u32 saved_regs_mpu[AZF_ALIGN(AZF_IO_SIZE_MPU_PM) / 4];
+ u32 saved_regs_opl3[AZF_ALIGN(AZF_IO_SIZE_OPL3_PM) / 4];
+ u32 saved_regs_mixer[AZF_ALIGN(AZF_IO_SIZE_MIXER_PM) / 4];
#endif
};
@@ -316,7 +360,7 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids);
static int
-snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set)
+snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set)
{
u8 prev = inb(reg), new;
@@ -331,39 +375,72 @@ snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set)
}
static inline void
-snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
+snd_azf3328_codec_outb(const struct snd_azf3328_codec_data *codec,
+ unsigned reg,
+ u8 value
+)
{
- outb(value, chip->codec_io + reg);
+ outb(value, codec->io_base + reg);
}
static inline u8
-snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg)
+snd_azf3328_codec_inb(const struct snd_azf3328_codec_data *codec, unsigned reg)
{
- return inb(chip->codec_io + reg);
+ return inb(codec->io_base + reg);
}
static inline void
-snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
+snd_azf3328_codec_outw(const struct snd_azf3328_codec_data *codec,
+ unsigned reg,
+ u16 value
+)
{
- outw(value, chip->codec_io + reg);
+ outw(value, codec->io_base + reg);
}
static inline u16
-snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg)
+snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg)
{
- return inw(chip->codec_io + reg);
+ return inw(codec->io_base + reg);
}
static inline void
-snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value)
+snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec,
+ unsigned reg,
+ u32 value
+)
{
- outl(value, chip->codec_io + reg);
+ outl(value, codec->io_base + reg);
}
static inline u32
-snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg)
+snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg)
+{
+ return inl(codec->io_base + reg);
+}
+
+static inline void
+snd_azf3328_ctrl_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
+{
+ outb(value, chip->ctrl_io + reg);
+}
+
+static inline u8
+snd_azf3328_ctrl_inb(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inb(chip->ctrl_io + reg);
+}
+
+static inline void
+snd_azf3328_ctrl_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
+{
+ outw(value, chip->ctrl_io + reg);
+}
+
+static inline void
+snd_azf3328_ctrl_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value)
{
- return inl(chip->codec_io + reg);
+ outl(value, chip->ctrl_io + reg);
}
static inline void
@@ -404,13 +481,13 @@ snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg)
#define AZF_MUTE_BIT 0x80
-static int
+static bool
snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip,
- unsigned reg, int do_mute
+ unsigned reg, bool do_mute
)
{
unsigned long portbase = chip->mixer_io + reg + 1;
- int updated;
+ bool updated;
/* the mute bit is on the *second* (i.e. right) register of a
* left/right channel setting */
@@ -569,7 +646,7 @@ snd_azf3328_get_mixer(struct snd_kcontrol *kcontrol,
{
struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
struct azf3328_mixer_reg reg;
- unsigned int oreg, val;
+ u16 oreg, val;
snd_azf3328_dbgcallenter();
snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
@@ -600,7 +677,7 @@ snd_azf3328_put_mixer(struct snd_kcontrol *kcontrol,
{
struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
struct azf3328_mixer_reg reg;
- unsigned int oreg, nreg, val;
+ u16 oreg, nreg, val;
snd_azf3328_dbgcallenter();
snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
@@ -709,7 +786,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol,
{
struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
struct azf3328_mixer_reg reg;
- unsigned int oreg, nreg, val;
+ u16 oreg, nreg, val;
snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
oreg = snd_azf3328_mixer_inw(chip, reg.reg);
@@ -867,14 +944,15 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream)
static void
snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
- unsigned reg,
+ enum snd_azf3328_codec_type codec_type,
enum azf_freq_t bitrate,
unsigned int format_width,
unsigned int channels
)
{
- u16 val = 0xff00;
unsigned long flags;
+ const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
+ u16 val = 0xff00;
snd_azf3328_dbgcallenter();
switch (bitrate) {
@@ -917,7 +995,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
spin_lock_irqsave(&chip->reg_lock, flags);
/* set bitrate/format */
- snd_azf3328_codec_outw(chip, reg, val);
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val);
/* changing the bitrate/format settings switches off the
* audio output with an annoying click in case of 8/16bit format change
@@ -926,11 +1004,11 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
* (FIXME: yes, it works, but what exactly am I doing here?? :)
* FIXME: does this have some side effects for full-duplex
* or other dramatic side effects? */
- if (reg == IDX_IO_PLAY_SOUNDFORMAT) /* only do it for playback */
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
- snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) |
- DMA_PLAY_SOMETHING1 |
- DMA_PLAY_SOMETHING2 |
+ if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
+ snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) |
+ DMA_RUN_SOMETHING1 |
+ DMA_RUN_SOMETHING2 |
SOMETHING_ALMOST_ALWAYS_SET |
DMA_EPILOGUE_SOMETHING |
DMA_SOMETHING_ELSE
@@ -942,112 +1020,134 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
static inline void
snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip,
- unsigned reg
+ enum snd_azf3328_codec_type codec_type
)
{
/* choose lowest frequency for low power consumption.
* While this will cause louder noise due to rather coarse frequency,
* it should never matter since output should always
* get disabled properly when idle anyway. */
- snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1);
+ snd_azf3328_codec_setfmt(chip, codec_type, AZF_FREQ_4000, 8, 1);
}
static void
-snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip,
+snd_azf3328_ctrl_reg_6AH_update(struct snd_azf3328 *chip,
unsigned bitmask,
- int enable
+ bool enable
)
{
- if (enable)
- chip->shadow_reg_codec_6AH &= ~bitmask;
+ bool do_mask = !enable;
+ if (do_mask)
+ chip->shadow_reg_ctrl_6AH |= bitmask;
else
- chip->shadow_reg_codec_6AH |= bitmask;
- snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n",
- bitmask, enable, chip->shadow_reg_codec_6AH);
- snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH);
+ chip->shadow_reg_ctrl_6AH &= ~bitmask;
+ snd_azf3328_dbgcodec("6AH_update mask 0x%04x do_mask %d: val 0x%04x\n",
+ bitmask, do_mask, chip->shadow_reg_ctrl_6AH);
+ snd_azf3328_ctrl_outw(chip, IDX_IO_6AH, chip->shadow_reg_ctrl_6AH);
}
static inline void
-snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable)
+snd_azf3328_ctrl_enable_codecs(struct snd_azf3328 *chip, bool enable)
{
- snd_azf3328_dbgplay("codec_enable %d\n", enable);
+ snd_azf3328_dbgcodec("codec_enable %d\n", enable);
/* no idea what exactly is being done here, but I strongly assume it's
* PM related */
- snd_azf3328_codec_reg_6AH_update(
+ snd_azf3328_ctrl_reg_6AH_update(
chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable
);
}
static void
-snd_azf3328_codec_activity(struct snd_azf3328 *chip,
- enum snd_azf3328_stream_index stream_type,
- int enable
+snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip,
+ enum snd_azf3328_codec_type codec_type,
+ bool enable
)
{
- int need_change = (chip->audio_stream[stream_type].running != enable);
+ struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
+ bool need_change = (codec->running != enable);
- snd_azf3328_dbgplay(
- "codec_activity: type %d, enable %d, need_change %d\n",
- stream_type, enable, need_change
+ snd_azf3328_dbgcodec(
+ "codec_activity: %s codec, enable %d, need_change %d\n",
+ codec->name, enable, need_change
);
if (need_change) {
- enum snd_azf3328_stream_index other =
- (stream_type == AZF_PLAYBACK) ?
- AZF_CAPTURE : AZF_PLAYBACK;
- /* small check to prevent shutting down the other party
- * in case it's active */
- if ((enable) || !(chip->audio_stream[other].running))
- snd_azf3328_codec_enable(chip, enable);
+ static const struct {
+ enum snd_azf3328_codec_type other1;
+ enum snd_azf3328_codec_type other2;
+ } peer_codecs[3] =
+ { { AZF_CODEC_CAPTURE, AZF_CODEC_I2S_OUT },
+ { AZF_CODEC_PLAYBACK, AZF_CODEC_I2S_OUT },
+ { AZF_CODEC_PLAYBACK, AZF_CODEC_CAPTURE } };
+ bool call_function;
+
+ if (enable)
+ /* if enable codec, call enable_codecs func
+ to enable codec supply... */
+ call_function = 1;
+ else {
+ /* ...otherwise call enable_codecs func
+ (which globally shuts down operation of codecs)
+ only in case the other codecs are currently
+ not active either! */
+ call_function =
+ ((!chip->codecs[peer_codecs[codec_type].other1]
+ .running)
+ && (!chip->codecs[peer_codecs[codec_type].other2]
+ .running));
+ }
+ if (call_function)
+ snd_azf3328_ctrl_enable_codecs(chip, enable);
/* ...and adjust clock, too
* (reduce noise and power consumption) */
if (!enable)
snd_azf3328_codec_setfmt_lowpower(
chip,
- chip->audio_stream[stream_type].portbase
- + IDX_IO_PLAY_SOUNDFORMAT
+ codec_type
);
+ codec->running = enable;
}
- chip->audio_stream[stream_type].running = enable;
}
static void
-snd_azf3328_setdmaa(struct snd_azf3328 *chip,
- long unsigned int addr,
- unsigned int count,
- unsigned int size,
- enum snd_azf3328_stream_index stream_type
+snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip,
+ enum snd_azf3328_codec_type codec_type,
+ unsigned long addr,
+ unsigned int count,
+ unsigned int size
)
{
+ const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
snd_azf3328_dbgcallenter();
- if (!chip->audio_stream[stream_type].running) {
- /* AZF3328 uses a two buffer pointer DMA playback approach */
+ if (!codec->running) {
+ /* AZF3328 uses a two buffer pointer DMA transfer approach */
- unsigned long flags, portbase, addr_area2;
+ unsigned long flags, addr_area2;
/* width 32bit (prevent overflow): */
- unsigned long count_areas, count_tmp;
+ u32 count_areas, lengths;
- portbase = chip->audio_stream[stream_type].portbase;
count_areas = size/2;
addr_area2 = addr+count_areas;
count_areas--; /* max. index */
- snd_azf3328_dbgplay("set DMA: buf1 %08lx[%lu], buf2 %08lx[%lu]\n", addr, count_areas, addr_area2, count_areas);
+ snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n",
+ addr, count_areas, addr_area2, count_areas);
/* build combined I/O buffer length word */
- count_tmp = count_areas;
- count_areas |= (count_tmp << 16);
+ lengths = (count_areas << 16) | (count_areas);
spin_lock_irqsave(&chip->reg_lock, flags);
- outl(addr, portbase + IDX_IO_PLAY_DMA_START_1);
- outl(addr_area2, portbase + IDX_IO_PLAY_DMA_START_2);
- outl(count_areas, portbase + IDX_IO_PLAY_DMA_LEN_1);
+ snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr);
+ snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2,
+ addr_area2);
+ snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS,
+ lengths);
spin_unlock_irqrestore(&chip->reg_lock, flags);
}
snd_azf3328_dbgcallleave();
}
static int
-snd_azf3328_playback_prepare(struct snd_pcm_substream *substream)
+snd_azf3328_codec_prepare(struct snd_pcm_substream *substream)
{
#if 0
struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
@@ -1058,157 +1158,161 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream)
snd_azf3328_dbgcallenter();
#if 0
- snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+ snd_azf3328_codec_setfmt(chip, AZF_CODEC_...,
runtime->rate,
snd_pcm_format_width(runtime->format),
runtime->channels);
- snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK);
+ snd_azf3328_codec_setdmaa(chip, AZF_CODEC_...,
+ runtime->dma_addr, count, size);
#endif
snd_azf3328_dbgcallleave();
return 0;
}
static int
-snd_azf3328_capture_prepare(struct snd_pcm_substream *substream)
-{
-#if 0
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- unsigned int size = snd_pcm_lib_buffer_bytes(substream);
- unsigned int count = snd_pcm_lib_period_bytes(substream);
-#endif
-
- snd_azf3328_dbgcallenter();
-#if 0
- snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
- runtime->rate,
- snd_pcm_format_width(runtime->format),
- runtime->channels);
- snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE);
-#endif
- snd_azf3328_dbgcallleave();
- return 0;
-}
-
-static int
-snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type,
+ struct snd_pcm_substream *substream, int cmd)
{
struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
struct snd_pcm_runtime *runtime = substream->runtime;
int result = 0;
- unsigned int status1;
- int previously_muted;
+ u16 flags1;
+ bool previously_muted = 0;
+ bool is_playback_codec = (AZF_CODEC_PLAYBACK == codec_type);
- snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd);
+ snd_azf3328_dbgcalls("snd_azf3328_codec_trigger cmd %d\n", cmd);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- snd_azf3328_dbgplay("START PLAYBACK\n");
-
- /* mute WaveOut (avoid clicking during setup) */
- previously_muted =
- snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+ snd_azf3328_dbgcodec("START %s\n", codec->name);
+
+ if (is_playback_codec) {
+ /* mute WaveOut (avoid clicking during setup) */
+ previously_muted =
+ snd_azf3328_mixer_set_mute(
+ chip, IDX_MIXER_WAVEOUT, 1
+ );
+ }
- snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+ snd_azf3328_codec_setfmt(chip, codec_type,
runtime->rate,
snd_pcm_format_width(runtime->format),
runtime->channels);
spin_lock(&chip->reg_lock);
/* first, remember current value: */
- status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
+ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS);
- /* stop playback */
- status1 &= ~DMA_RESUME;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ /* stop transfer */
+ flags1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
/* FIXME: clear interrupts or what??? */
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff);
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_IRQTYPE, 0xffff);
spin_unlock(&chip->reg_lock);
- snd_azf3328_setdmaa(chip, runtime->dma_addr,
+ snd_azf3328_codec_setdmaa(chip, codec_type, runtime->dma_addr,
snd_pcm_lib_period_bytes(substream),
- snd_pcm_lib_buffer_bytes(substream),
- AZF_PLAYBACK);
+ snd_pcm_lib_buffer_bytes(substream)
+ );
spin_lock(&chip->reg_lock);
#ifdef WIN9X
/* FIXME: enable playback/recording??? */
- status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ flags1 |= DMA_RUN_SOMETHING1 | DMA_RUN_SOMETHING2;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
- /* start playback again */
+ /* start transfer again */
/* FIXME: what is this value (0x0010)??? */
- status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ flags1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
#else /* NT4 */
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
0x0000);
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
- DMA_PLAY_SOMETHING1);
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
- DMA_PLAY_SOMETHING1 |
- DMA_PLAY_SOMETHING2);
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
+ DMA_RUN_SOMETHING1);
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
+ DMA_RUN_SOMETHING1 |
+ DMA_RUN_SOMETHING2);
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
DMA_RESUME |
SOMETHING_ALMOST_ALWAYS_SET |
DMA_EPILOGUE_SOMETHING |
DMA_SOMETHING_ELSE);
#endif
spin_unlock(&chip->reg_lock);
- snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1);
-
- /* now unmute WaveOut */
- if (!previously_muted)
- snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+ snd_azf3328_ctrl_codec_activity(chip, codec_type, 1);
+
+ if (is_playback_codec) {
+ /* now unmute WaveOut */
+ if (!previously_muted)
+ snd_azf3328_mixer_set_mute(
+ chip, IDX_MIXER_WAVEOUT, 0
+ );
+ }
- snd_azf3328_dbgplay("STARTED PLAYBACK\n");
+ snd_azf3328_dbgcodec("STARTED %s\n", codec->name);
break;
case SNDRV_PCM_TRIGGER_RESUME:
- snd_azf3328_dbgplay("RESUME PLAYBACK\n");
- /* resume playback if we were active */
+ snd_azf3328_dbgcodec("RESUME %s\n", codec->name);
+ /* resume codec if we were active */
spin_lock(&chip->reg_lock);
- if (chip->audio_stream[AZF_PLAYBACK].running)
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
- snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME);
+ if (codec->running)
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
+ snd_azf3328_codec_inw(
+ codec, IDX_IO_CODEC_DMA_FLAGS
+ ) | DMA_RESUME
+ );
spin_unlock(&chip->reg_lock);
break;
case SNDRV_PCM_TRIGGER_STOP:
- snd_azf3328_dbgplay("STOP PLAYBACK\n");
-
- /* mute WaveOut (avoid clicking during setup) */
- previously_muted =
- snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+ snd_azf3328_dbgcodec("STOP %s\n", codec->name);
+
+ if (is_playback_codec) {
+ /* mute WaveOut (avoid clicking during setup) */
+ previously_muted =
+ snd_azf3328_mixer_set_mute(
+ chip, IDX_MIXER_WAVEOUT, 1
+ );
+ }
spin_lock(&chip->reg_lock);
/* first, remember current value: */
- status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
+ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS);
- /* stop playback */
- status1 &= ~DMA_RESUME;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ /* stop transfer */
+ flags1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
/* hmm, is this really required? we're resetting the same bit
* immediately thereafter... */
- status1 |= DMA_PLAY_SOMETHING1;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ flags1 |= DMA_RUN_SOMETHING1;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
- status1 &= ~DMA_PLAY_SOMETHING1;
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ flags1 &= ~DMA_RUN_SOMETHING1;
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1);
spin_unlock(&chip->reg_lock);
- snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
-
- /* now unmute WaveOut */
- if (!previously_muted)
- snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+ snd_azf3328_ctrl_codec_activity(chip, codec_type, 0);
+
+ if (is_playback_codec) {
+ /* now unmute WaveOut */
+ if (!previously_muted)
+ snd_azf3328_mixer_set_mute(
+ chip, IDX_MIXER_WAVEOUT, 0
+ );
+ }
- snd_azf3328_dbgplay("STOPPED PLAYBACK\n");
+ snd_azf3328_dbgcodec("STOPPED %s\n", codec->name);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
- snd_azf3328_dbgplay("SUSPEND PLAYBACK\n");
- /* make sure playback is stopped */
- snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
- snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME);
+ snd_azf3328_dbgcodec("SUSPEND %s\n", codec->name);
+ /* make sure codec is stopped */
+ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
+ snd_azf3328_codec_inw(
+ codec, IDX_IO_CODEC_DMA_FLAGS
+ ) & ~DMA_RESUME
+ );
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
@@ -1217,7 +1321,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
break;
default:
- printk(KERN_ERR "FIXME: unknown trigger mode!\n");
+ snd_printk(KERN_ERR "FIXME: unknown trigger mode!\n");
return -EINVAL;
}
@@ -1225,172 +1329,74 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
return result;
}
-/* this is just analogous to playback; I'm not quite sure whether recording
- * should actually be triggered like that */
static int
-snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+snd_azf3328_codec_playback_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
- int result = 0;
- unsigned int status1;
-
- snd_azf3328_dbgcalls("snd_azf3328_capture_trigger cmd %d\n", cmd);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
-
- snd_azf3328_dbgplay("START CAPTURE\n");
-
- snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
- runtime->rate,
- snd_pcm_format_width(runtime->format),
- runtime->channels);
-
- spin_lock(&chip->reg_lock);
- /* first, remember current value: */
- status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
-
- /* stop recording */
- status1 &= ~DMA_RESUME;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-
- /* FIXME: clear interrupts or what??? */
- snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff);
- spin_unlock(&chip->reg_lock);
-
- snd_azf3328_setdmaa(chip, runtime->dma_addr,
- snd_pcm_lib_period_bytes(substream),
- snd_pcm_lib_buffer_bytes(substream),
- AZF_CAPTURE);
-
- spin_lock(&chip->reg_lock);
-#ifdef WIN9X
- /* FIXME: enable playback/recording??? */
- status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-
- /* start capture again */
- /* FIXME: what is this value (0x0010)??? */
- status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-#else
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- 0x0000);
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- DMA_PLAY_SOMETHING1);
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- DMA_PLAY_SOMETHING1 |
- DMA_PLAY_SOMETHING2);
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- DMA_RESUME |
- SOMETHING_ALMOST_ALWAYS_SET |
- DMA_EPILOGUE_SOMETHING |
- DMA_SOMETHING_ELSE);
-#endif
- spin_unlock(&chip->reg_lock);
- snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1);
-
- snd_azf3328_dbgplay("STARTED CAPTURE\n");
- break;
- case SNDRV_PCM_TRIGGER_RESUME:
- snd_azf3328_dbgplay("RESUME CAPTURE\n");
- /* resume recording if we were active */
- spin_lock(&chip->reg_lock);
- if (chip->audio_stream[AZF_CAPTURE].running)
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME);
- spin_unlock(&chip->reg_lock);
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- snd_azf3328_dbgplay("STOP CAPTURE\n");
-
- spin_lock(&chip->reg_lock);
- /* first, remember current value: */
- status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
-
- /* stop recording */
- status1 &= ~DMA_RESUME;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-
- status1 |= DMA_PLAY_SOMETHING1;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
-
- status1 &= ~DMA_PLAY_SOMETHING1;
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
- spin_unlock(&chip->reg_lock);
- snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0);
+ return snd_azf3328_codec_trigger(AZF_CODEC_PLAYBACK, substream, cmd);
+}
- snd_azf3328_dbgplay("STOPPED CAPTURE\n");
- break;
- case SNDRV_PCM_TRIGGER_SUSPEND:
- snd_azf3328_dbgplay("SUSPEND CAPTURE\n");
- /* make sure recording is stopped */
- snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
- snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME);
- break;
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
- break;
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
- break;
- default:
- printk(KERN_ERR "FIXME: unknown trigger mode!\n");
- return -EINVAL;
- }
+static int
+snd_azf3328_codec_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ return snd_azf3328_codec_trigger(AZF_CODEC_CAPTURE, substream, cmd);
+}
- snd_azf3328_dbgcallleave();
- return result;
+static int
+snd_azf3328_codec_i2s_out_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ return snd_azf3328_codec_trigger(AZF_CODEC_I2S_OUT, substream, cmd);
}
static snd_pcm_uframes_t
-snd_azf3328_playback_pointer(struct snd_pcm_substream *substream)
+snd_azf3328_codec_pointer(struct snd_pcm_substream *substream,
+ enum snd_azf3328_codec_type codec_type
+)
{
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
unsigned long bufptr, result;
snd_pcm_uframes_t frmres;
#ifdef QUERY_HARDWARE
- bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1);
+ bufptr = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1);
#else
bufptr = substream->runtime->dma_addr;
#endif
- result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS);
+ result = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_CURRPOS);
/* calculate offset */
result -= bufptr;
frmres = bytes_to_frames( substream->runtime, result);
- snd_azf3328_dbgplay("PLAY @ 0x%8lx, frames %8ld\n", result, frmres);
+ snd_azf3328_dbgcodec("%s @ 0x%8lx, frames %8ld\n",
+ codec->name, result, frmres);
return frmres;
}
static snd_pcm_uframes_t
-snd_azf3328_capture_pointer(struct snd_pcm_substream *substream)
+snd_azf3328_codec_playback_pointer(struct snd_pcm_substream *substream)
{
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
- unsigned long bufptr, result;
- snd_pcm_uframes_t frmres;
+ return snd_azf3328_codec_pointer(substream, AZF_CODEC_PLAYBACK);
+}
-#ifdef QUERY_HARDWARE
- bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1);
-#else
- bufptr = substream->runtime->dma_addr;
-#endif
- result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS);
+static snd_pcm_uframes_t
+snd_azf3328_codec_capture_pointer(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_codec_pointer(substream, AZF_CODEC_CAPTURE);
+}
- /* calculate offset */
- result -= bufptr;
- frmres = bytes_to_frames( substream->runtime, result);
- snd_azf3328_dbgplay("REC @ 0x%8lx, frames %8ld\n", result, frmres);
- return frmres;
+static snd_pcm_uframes_t
+snd_azf3328_codec_i2s_out_pointer(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_codec_pointer(substream, AZF_CODEC_I2S_OUT);
}
/******************************************************************/
#ifdef SUPPORT_GAMEPORT
static inline void
-snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable)
+snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip,
+ bool enable
+)
{
snd_azf3328_io_reg_setb(
chip->game_io+IDX_GAME_HWCONFIG,
@@ -1400,7 +1406,9 @@ snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable)
}
static inline void
-snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable)
+snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip,
+ bool enable
+)
{
snd_azf3328_io_reg_setb(
chip->game_io+IDX_GAME_HWCONFIG,
@@ -1409,10 +1417,27 @@ snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable)
);
}
+static void
+snd_azf3328_gameport_set_counter_frequency(struct snd_azf3328 *chip,
+ unsigned int freq_cfg
+)
+{
+ snd_azf3328_io_reg_setb(
+ chip->game_io+IDX_GAME_HWCONFIG,
+ 0x02,
+ (freq_cfg & 1) != 0
+ );
+ snd_azf3328_io_reg_setb(
+ chip->game_io+IDX_GAME_HWCONFIG,
+ 0x04,
+ (freq_cfg & 2) != 0
+ );
+}
+
static inline void
-snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable)
+snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, bool enable)
{
- snd_azf3328_codec_reg_6AH_update(
+ snd_azf3328_ctrl_reg_6AH_update(
chip, IO_6A_SOMETHING2_GAMEPORT, enable
);
}
@@ -1447,6 +1472,8 @@ snd_azf3328_gameport_open(struct gameport *gameport, int mode)
break;
}
+ snd_azf3328_gameport_set_counter_frequency(chip,
+ GAME_HWCFG_ADC_COUNTER_FREQ_STD);
snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0));
return res;
@@ -1458,6 +1485,8 @@ snd_azf3328_gameport_close(struct gameport *gameport)
struct snd_azf3328 *chip = gameport_get_port_data(gameport);
snd_azf3328_dbggame("gameport_close\n");
+ snd_azf3328_gameport_set_counter_frequency(chip,
+ GAME_HWCFG_ADC_COUNTER_FREQ_1_200);
snd_azf3328_gameport_axis_circuit_enable(chip, 0);
}
@@ -1491,7 +1520,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport,
val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG);
if (val & GAME_AXES_SAMPLING_READY) {
- for (i = 0; i < 4; ++i) {
+ for (i = 0; i < ARRAY_SIZE(chip->axes); ++i) {
/* configure the axis to read */
val = (i << 4) | 0x0f;
snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val);
@@ -1514,7 +1543,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport,
snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff);
spin_unlock_irqrestore(&chip->reg_lock, flags);
- for (i = 0; i < 4; i++) {
+ for (i = 0; i < ARRAY_SIZE(chip->axes); i++) {
axes[i] = chip->axes[i];
if (axes[i] == 0xffff)
axes[i] = -1;
@@ -1552,6 +1581,8 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev)
/* DISABLE legacy address: we don't need it! */
snd_azf3328_gameport_legacy_address_enable(chip, 0);
+ snd_azf3328_gameport_set_counter_frequency(chip,
+ GAME_HWCFG_ADC_COUNTER_FREQ_1_200);
snd_azf3328_gameport_axis_circuit_enable(chip, 0);
gameport_register_port(chip->gameport);
@@ -1585,40 +1616,77 @@ snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip)
static inline void
snd_azf3328_irq_log_unknown_type(u8 which)
{
- snd_azf3328_dbgplay(
+ snd_azf3328_dbgcodec(
"azt3328: unknown IRQ type (%x) occurred, please report!\n",
which
);
}
+static inline void
+snd_azf3328_codec_interrupt(struct snd_azf3328 *chip, u8 status)
+{
+ u8 which;
+ enum snd_azf3328_codec_type codec_type;
+ const struct snd_azf3328_codec_data *codec;
+
+ for (codec_type = AZF_CODEC_PLAYBACK;
+ codec_type <= AZF_CODEC_I2S_OUT;
+ ++codec_type) {
+
+ /* skip codec if there's no interrupt for it */
+ if (!(status & (1 << codec_type)))
+ continue;
+
+ codec = &chip->codecs[codec_type];
+
+ spin_lock(&chip->reg_lock);
+ which = snd_azf3328_codec_inb(codec, IDX_IO_CODEC_IRQTYPE);
+ /* ack all IRQ types immediately */
+ snd_azf3328_codec_outb(codec, IDX_IO_CODEC_IRQTYPE, which);
+ spin_unlock(&chip->reg_lock);
+
+ if ((chip->pcm[codec_type]) && (codec->substream)) {
+ snd_pcm_period_elapsed(codec->substream);
+ snd_azf3328_dbgcodec("%s period done (#%x), @ %x\n",
+ codec->name,
+ which,
+ snd_azf3328_codec_inl(
+ codec, IDX_IO_CODEC_DMA_CURRPOS
+ )
+ );
+ } else
+ printk(KERN_WARNING "azt3328: irq handler problem!\n");
+ if (which & IRQ_SOMETHING)
+ snd_azf3328_irq_log_unknown_type(which);
+ }
+}
+
static irqreturn_t
snd_azf3328_interrupt(int irq, void *dev_id)
{
struct snd_azf3328 *chip = dev_id;
- u8 status, which;
-#if DEBUG_PLAY_REC
+ u8 status;
+#if DEBUG_CODEC
static unsigned long irq_count;
#endif
- status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS);
+ status = snd_azf3328_ctrl_inb(chip, IDX_IO_IRQSTATUS);
/* fast path out, to ease interrupt sharing */
if (!(status &
- (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER)
+ (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT
+ |IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER)
))
return IRQ_NONE; /* must be interrupt for another device */
- snd_azf3328_dbgplay(
- "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, "
- "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n",
+ snd_azf3328_dbgcodec(
+ "irq_count %ld! IDX_IO_IRQSTATUS %04x\n",
irq_count++ /* debug-only */,
- snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS),
- snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE),
status
);
if (status & IRQ_TIMER) {
- /* snd_azf3328_dbgplay("timer %ld\n",
+ /* snd_azf3328_dbgcodec("timer %ld\n",
snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE)
& TIMER_VALUE_MASK
); */
@@ -1626,71 +1694,36 @@ snd_azf3328_interrupt(int irq, void *dev_id)
snd_timer_interrupt(chip->timer, chip->timer->sticks);
/* ACK timer */
spin_lock(&chip->reg_lock);
- snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07);
+ snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07);
spin_unlock(&chip->reg_lock);
- snd_azf3328_dbgplay("azt3328: timer IRQ\n");
+ snd_azf3328_dbgcodec("azt3328: timer IRQ\n");
}
- if (status & IRQ_PLAYBACK) {
- spin_lock(&chip->reg_lock);
- which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE);
- /* ack all IRQ types immediately */
- snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which);
- spin_unlock(&chip->reg_lock);
- if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) {
- snd_pcm_period_elapsed(
- chip->audio_stream[AZF_PLAYBACK].substream
- );
- snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n",
- which,
- snd_azf3328_codec_inl(
- chip, IDX_IO_PLAY_DMA_CURRPOS
- )
- );
- } else
- printk(KERN_WARNING "azt3328: irq handler problem!\n");
- if (which & IRQ_PLAY_SOMETHING)
- snd_azf3328_irq_log_unknown_type(which);
- }
- if (status & IRQ_RECORDING) {
- spin_lock(&chip->reg_lock);
- which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE);
- /* ack all IRQ types immediately */
- snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which);
- spin_unlock(&chip->reg_lock);
+ if (status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT))
+ snd_azf3328_codec_interrupt(chip, status);
- if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) {
- snd_pcm_period_elapsed(
- chip->audio_stream[AZF_CAPTURE].substream
- );
- snd_azf3328_dbgplay("REC period done (#%x), @ %x\n",
- which,
- snd_azf3328_codec_inl(
- chip, IDX_IO_REC_DMA_CURRPOS
- )
- );
- } else
- printk(KERN_WARNING "azt3328: irq handler problem!\n");
- if (which & IRQ_REC_SOMETHING)
- snd_azf3328_irq_log_unknown_type(which);
- }
if (status & IRQ_GAMEPORT)
snd_azf3328_gameport_interrupt(chip);
+
/* MPU401 has less critical IRQ requirements
* than timer and playback/recording, right? */
if (status & IRQ_MPU401) {
snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
/* hmm, do we have to ack the IRQ here somehow?
- * If so, then I don't know how... */
- snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n");
+ * If so, then I don't know how yet... */
+ snd_azf3328_dbgcodec("azt3328: MPU401 IRQ\n");
}
return IRQ_HANDLED;
}
/*****************************************************************/
-static const struct snd_pcm_hardware snd_azf3328_playback =
+/* as long as we think we have identical snd_pcm_hardware parameters
+ for playback, capture and i2s out, we can use the same physical struct
+ since the struct is simply being copied into a member.
+*/
+static const struct snd_pcm_hardware snd_azf3328_hardware =
{
/* FIXME!! Correct? */
.info = SNDRV_PCM_INFO_MMAP |
@@ -1718,31 +1751,6 @@ static const struct snd_pcm_hardware snd_azf3328_playback =
.fifo_size = 0,
};
-static const struct snd_pcm_hardware snd_azf3328_capture =
-{
- /* FIXME */
- .info = SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID,
- .formats = SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_U8 |
- SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_U16_LE,
- .rates = SNDRV_PCM_RATE_5512 |
- SNDRV_PCM_RATE_8000_48000 |
- SNDRV_PCM_RATE_KNOT,
- .rate_min = AZF_FREQ_4000,
- .rate_max = AZF_FREQ_66200,
- .channels_min = 1,
- .channels_max = 2,
- .buffer_bytes_max = 65536,
- .period_bytes_min = 64,
- .period_bytes_max = 65536,
- .periods_min = 1,
- .periods_max = 1024,
- .fifo_size = 0,
-};
-
static unsigned int snd_azf3328_fixed_rates[] = {
AZF_FREQ_4000,
@@ -1770,14 +1778,19 @@ static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = {
/*****************************************************************/
static int
-snd_azf3328_playback_open(struct snd_pcm_substream *substream)
+snd_azf3328_pcm_open(struct snd_pcm_substream *substream,
+ enum snd_azf3328_codec_type codec_type
+)
{
struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
snd_azf3328_dbgcallenter();
- chip->audio_stream[AZF_PLAYBACK].substream = substream;
- runtime->hw = snd_azf3328_playback;
+ chip->codecs[codec_type].substream = substream;
+
+ /* same parameters for all our codecs - at least we think so... */
+ runtime->hw = snd_azf3328_hardware;
+
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&snd_azf3328_hw_constraints_rates);
snd_azf3328_dbgcallleave();
@@ -1785,40 +1798,52 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream)
}
static int
+snd_azf3328_playback_open(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_pcm_open(substream, AZF_CODEC_PLAYBACK);
+}
+
+static int
snd_azf3328_capture_open(struct snd_pcm_substream *substream)
{
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
+ return snd_azf3328_pcm_open(substream, AZF_CODEC_CAPTURE);
+}
- snd_azf3328_dbgcallenter();
- chip->audio_stream[AZF_CAPTURE].substream = substream;
- runtime->hw = snd_azf3328_capture;
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
- &snd_azf3328_hw_constraints_rates);
- snd_azf3328_dbgcallleave();
- return 0;
+static int
+snd_azf3328_i2s_out_open(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_pcm_open(substream, AZF_CODEC_I2S_OUT);
}
static int
-snd_azf3328_playback_close(struct snd_pcm_substream *substream)
+snd_azf3328_pcm_close(struct snd_pcm_substream *substream,
+ enum snd_azf3328_codec_type codec_type
+)
{
struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
snd_azf3328_dbgcallenter();
- chip->audio_stream[AZF_PLAYBACK].substream = NULL;
+ chip->codecs[codec_type].substream = NULL;
snd_azf3328_dbgcallleave();
return 0;
}
static int
+snd_azf3328_playback_close(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_pcm_close(substream, AZF_CODEC_PLAYBACK);
+}
+
+static int
snd_azf3328_capture_close(struct snd_pcm_substream *substream)
{
- struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ return snd_azf3328_pcm_close(substream, AZF_CODEC_CAPTURE);
+}
- snd_azf3328_dbgcallenter();
- chip->audio_stream[AZF_CAPTURE].substream = NULL;
- snd_azf3328_dbgcallleave();
- return 0;
+static int
+snd_azf3328_i2s_out_close(struct snd_pcm_substream *substream)
+{
+ return snd_azf3328_pcm_close(substream, AZF_CODEC_I2S_OUT);
}
/******************************************************************/
@@ -1829,9 +1854,9 @@ static struct snd_pcm_ops snd_azf3328_playback_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_azf3328_hw_params,
.hw_free = snd_azf3328_hw_free,
- .prepare = snd_azf3328_playback_prepare,
- .trigger = snd_azf3328_playback_trigger,
- .pointer = snd_azf3328_playback_pointer
+ .prepare = snd_azf3328_codec_prepare,
+ .trigger = snd_azf3328_codec_playback_trigger,
+ .pointer = snd_azf3328_codec_playback_pointer
};
static struct snd_pcm_ops snd_azf3328_capture_ops = {
@@ -1840,30 +1865,67 @@ static struct snd_pcm_ops snd_azf3328_capture_ops = {
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_azf3328_hw_params,
.hw_free = snd_azf3328_hw_free,
- .prepare = snd_azf3328_capture_prepare,
- .trigger = snd_azf3328_capture_trigger,
- .pointer = snd_azf3328_capture_pointer
+ .prepare = snd_azf3328_codec_prepare,
+ .trigger = snd_azf3328_codec_capture_trigger,
+ .pointer = snd_azf3328_codec_capture_pointer
+};
+
+static struct snd_pcm_ops snd_azf3328_i2s_out_ops = {
+ .open = snd_azf3328_i2s_out_open,
+ .close = snd_azf3328_i2s_out_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_azf3328_hw_params,
+ .hw_free = snd_azf3328_hw_free,
+ .prepare = snd_azf3328_codec_prepare,
+ .trigger = snd_azf3328_codec_i2s_out_trigger,
+ .pointer = snd_azf3328_codec_i2s_out_pointer
};
static int __devinit
-snd_azf3328_pcm(struct snd_azf3328 *chip, int device)
+snd_azf3328_pcm(struct snd_azf3328 *chip)
{
+enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */
+
struct snd_pcm *pcm;
int err;
snd_azf3328_dbgcallenter();
- if ((err = snd_pcm_new(chip->card, "AZF3328 DSP", device, 1, 1, &pcm)) < 0)
+
+ err = snd_pcm_new(chip->card, "AZF3328 DSP", AZF_PCMDEV_STD,
+ 1, 1, &pcm);
+ if (err < 0)
return err;
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_azf3328_playback_ops);
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_azf3328_capture_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_azf3328_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_azf3328_capture_ops);
pcm->private_data = chip;
pcm->info_flags = 0;
strcpy(pcm->name, chip->card->shortname);
- chip->pcm = pcm;
+ /* same pcm object for playback/capture (see snd_pcm_new() above) */
+ chip->pcm[AZF_CODEC_PLAYBACK] = pcm;
+ chip->pcm[AZF_CODEC_CAPTURE] = pcm;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
- snd_dma_pci_data(chip->pci), 64*1024, 64*1024);
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 64*1024);
+
+ err = snd_pcm_new(chip->card, "AZF3328 I2S OUT", AZF_PCMDEV_I2S_OUT,
+ 1, 0, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_azf3328_i2s_out_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm[AZF_CODEC_I2S_OUT] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 64*1024);
snd_azf3328_dbgcallleave();
return 0;
@@ -1902,7 +1964,7 @@ snd_azf3328_timer_start(struct snd_timer *timer)
snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay);
delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE;
spin_lock_irqsave(&chip->reg_lock, flags);
- snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay);
+ snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay);
spin_unlock_irqrestore(&chip->reg_lock, flags);
snd_azf3328_dbgcallleave();
return 0;
@@ -1919,7 +1981,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer)
spin_lock_irqsave(&chip->reg_lock, flags);
/* disable timer countdown and interrupt */
/* FIXME: should we write TIMER_IRQ_ACK here? */
- snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
+ snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
spin_unlock_irqrestore(&chip->reg_lock, flags);
snd_azf3328_dbgcallleave();
return 0;
@@ -2035,7 +2097,7 @@ snd_azf3328_test_bit(unsigned unsigned reg, int bit)
outb(val, reg);
- printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n",
+ printk(KERN_DEBUG "reg %04x bit %d: %02x %02x %02x\n",
reg, bit, val, valoff, valon
);
}
@@ -2048,9 +2110,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip)
u16 tmp;
snd_azf3328_dbgmisc(
- "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, "
+ "ctrl_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, "
"opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n",
- chip->codec_io, chip->game_io, chip->mpu_io,
+ chip->ctrl_io, chip->game_io, chip->mpu_io,
chip->opl3_io, chip->mixer_io, chip->irq
);
@@ -2083,9 +2145,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip)
inb(0x38c + tmp)
);
- for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2)
- snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n",
- tmp, snd_azf3328_codec_inw(chip, tmp)
+ for (tmp = 0; tmp < AZF_IO_SIZE_CTRL; tmp += 2)
+ snd_azf3328_dbgmisc("ctrl 0x%02x: 0x%04x\n",
+ tmp, snd_azf3328_ctrl_inw(chip, tmp)
);
for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2)
@@ -2106,7 +2168,8 @@ snd_azf3328_create(struct snd_card *card,
static struct snd_device_ops ops = {
.dev_free = snd_azf3328_dev_free,
};
- u16 tmp;
+ u8 dma_init;
+ enum snd_azf3328_codec_type codec_type;
*rchip = NULL;
@@ -2138,14 +2201,21 @@ snd_azf3328_create(struct snd_card *card,
if (err < 0)
goto out_err;
- chip->codec_io = pci_resource_start(pci, 0);
+ chip->ctrl_io = pci_resource_start(pci, 0);
chip->game_io = pci_resource_start(pci, 1);
chip->mpu_io = pci_resource_start(pci, 2);
- chip->opl3_io = pci_resource_start(pci, 3);
+ chip->opl3_io = pci_resource_start(pci, 3);
chip->mixer_io = pci_resource_start(pci, 4);
- chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00;
- chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20;
+ chip->codecs[AZF_CODEC_PLAYBACK].io_base =
+ chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK;
+ chip->codecs[AZF_CODEC_PLAYBACK].name = "PLAYBACK";
+ chip->codecs[AZF_CODEC_CAPTURE].io_base =
+ chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE;
+ chip->codecs[AZF_CODEC_CAPTURE].name = "CAPTURE";
+ chip->codecs[AZF_CODEC_I2S_OUT].io_base =
+ chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT;
+ chip->codecs[AZF_CODEC_I2S_OUT].name = "I2S_OUT";
if (request_irq(pci->irq, snd_azf3328_interrupt,
IRQF_SHARED, card->shortname, chip)) {
@@ -2168,20 +2238,25 @@ snd_azf3328_create(struct snd_card *card,
if (err < 0)
goto out_err;
- /* shutdown codecs to save power */
- /* have snd_azf3328_codec_activity() act properly */
- chip->audio_stream[AZF_PLAYBACK].running = 1;
- snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
+ /* standard codec init stuff */
+ /* default DMA init value */
+ dma_init = DMA_RUN_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE;
- /* standard chip init stuff */
- /* default IRQ init value */
- tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE;
+ for (codec_type = AZF_CODEC_PLAYBACK;
+ codec_type <= AZF_CODEC_I2S_OUT; ++codec_type) {
+ struct snd_azf3328_codec_data *codec =
+ &chip->codecs[codec_type];
- spin_lock_irq(&chip->reg_lock);
- snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp);
- snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp);
- snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp);
- spin_unlock_irq(&chip->reg_lock);
+ /* shutdown codecs to save power */
+ /* have ...ctrl_codec_activity() act properly */
+ codec->running = 1;
+ snd_azf3328_ctrl_codec_activity(chip, codec_type, 0);
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_azf3328_codec_outb(codec, IDX_IO_CODEC_DMA_FLAGS,
+ dma_init);
+ spin_unlock_irq(&chip->reg_lock);
+ }
snd_card_set_dev(card, &pci->dev);
@@ -2229,8 +2304,11 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
card->private_data = chip;
+ /* chose to use MPU401_HW_AZT2320 ID instead of MPU401_HW_MPU401,
+ since our hardware ought to be similar, thus use same ID. */
err = snd_mpu401_uart_new(
- card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED,
+ card, 0,
+ MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED,
pci->irq, 0, &chip->rmidi
);
if (err < 0) {
@@ -2244,7 +2322,7 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
if (err < 0)
goto out_err;
- err = snd_azf3328_pcm(chip, 0);
+ err = snd_azf3328_pcm(chip);
if (err < 0)
goto out_err;
@@ -2266,14 +2344,14 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
opl3->private_data = chip;
sprintf(card->longname, "%s at 0x%lx, irq %i",
- card->shortname, chip->codec_io, chip->irq);
+ card->shortname, chip->ctrl_io, chip->irq);
err = snd_card_register(card);
if (err < 0)
goto out_err;
#ifdef MODULE
- printk(
+ printk(KERN_INFO
"azt3328: Sound driver for Aztech AZF3328-based soundcards such as PCI168.\n"
"azt3328: Hardware was completely undocumented, unfortunately.\n"
"azt3328: Feel free to contact andi AT lisas.de for bug reports etc.!\n"
@@ -2308,36 +2386,52 @@ snd_azf3328_remove(struct pci_dev *pci)
}
#ifdef CONFIG_PM
+static inline void
+snd_azf3328_suspend_regs(unsigned long io_addr, unsigned count, u32 *saved_regs)
+{
+ unsigned reg;
+
+ for (reg = 0; reg < count; ++reg) {
+ *saved_regs = inl(io_addr);
+ snd_azf3328_dbgpm("suspend: io 0x%04lx: 0x%08x\n",
+ io_addr, *saved_regs);
+ ++saved_regs;
+ io_addr += sizeof(*saved_regs);
+ }
+}
+
static int
snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
struct snd_azf3328 *chip = card->private_data;
- unsigned reg;
+ u16 *saved_regs_ctrl_u16;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
- snd_pcm_suspend_all(chip->pcm);
+ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]);
+ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]);
- for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
- chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2);
+ snd_azf3328_suspend_regs(chip->mixer_io,
+ ARRAY_SIZE(chip->saved_regs_mixer), chip->saved_regs_mixer);
/* make sure to disable master volume etc. to prevent looping sound */
snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
- for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
- chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2);
+ snd_azf3328_suspend_regs(chip->ctrl_io,
+ ARRAY_SIZE(chip->saved_regs_ctrl), chip->saved_regs_ctrl);
/* manually store the one currently relevant write-only reg, too */
- chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH;
+ saved_regs_ctrl_u16 = (u16 *)chip->saved_regs_ctrl;
+ saved_regs_ctrl_u16[IDX_IO_6AH / 2] = chip->shadow_reg_ctrl_6AH;
- for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
- chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
- chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
- chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2);
+ snd_azf3328_suspend_regs(chip->game_io,
+ ARRAY_SIZE(chip->saved_regs_game), chip->saved_regs_game);
+ snd_azf3328_suspend_regs(chip->mpu_io,
+ ARRAY_SIZE(chip->saved_regs_mpu), chip->saved_regs_mpu);
+ snd_azf3328_suspend_regs(chip->opl3_io,
+ ARRAY_SIZE(chip->saved_regs_opl3), chip->saved_regs_opl3);
pci_disable_device(pci);
pci_save_state(pci);
@@ -2345,12 +2439,28 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
return 0;
}
+static inline void
+snd_azf3328_resume_regs(const u32 *saved_regs,
+ unsigned long io_addr,
+ unsigned count
+)
+{
+ unsigned reg;
+
+ for (reg = 0; reg < count; ++reg) {
+ outl(*saved_regs, io_addr);
+ snd_azf3328_dbgpm("resume: io 0x%04lx: 0x%08x --> 0x%08x\n",
+ io_addr, *saved_regs, inl(io_addr));
+ ++saved_regs;
+ io_addr += sizeof(*saved_regs);
+ }
+}
+
static int
snd_azf3328_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
- struct snd_azf3328 *chip = card->private_data;
- unsigned reg;
+ const struct snd_azf3328 *chip = card->private_data;
pci_set_power_state(pci, PCI_D0);
pci_restore_state(pci);
@@ -2362,16 +2472,24 @@ snd_azf3328_resume(struct pci_dev *pci)
}
pci_set_master(pci);
- for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
- outw(chip->saved_regs_game[reg], chip->game_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
- outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
- outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
- outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2);
- for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
- outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2);
+ snd_azf3328_resume_regs(chip->saved_regs_game, chip->game_io,
+ ARRAY_SIZE(chip->saved_regs_game));
+ snd_azf3328_resume_regs(chip->saved_regs_mpu, chip->mpu_io,
+ ARRAY_SIZE(chip->saved_regs_mpu));
+ snd_azf3328_resume_regs(chip->saved_regs_opl3, chip->opl3_io,
+ ARRAY_SIZE(chip->saved_regs_opl3));
+
+ snd_azf3328_resume_regs(chip->saved_regs_mixer, chip->mixer_io,
+ ARRAY_SIZE(chip->saved_regs_mixer));
+
+ /* unfortunately with 32bit transfers, IDX_MIXER_PLAY_MASTER (0x02)
+ and IDX_MIXER_RESET (offset 0x00) get touched at the same time,
+ resulting in a mixer reset condition persisting until _after_
+ master vol was restored. Thus master vol needs an extra restore. */
+ outw(((u16 *)chip->saved_regs_mixer)[1], chip->mixer_io + 2);
+
+ snd_azf3328_resume_regs(chip->saved_regs_ctrl, chip->ctrl_io,
+ ARRAY_SIZE(chip->saved_regs_ctrl));
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h
index 974e051..6f46b97 100644
--- a/sound/pci/azt3328.h
+++ b/sound/pci/azt3328.h
@@ -6,50 +6,59 @@
/*** main I/O area port indices ***/
/* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */
-#define AZF_IO_SIZE_CODEC 0x80
-#define AZF_IO_SIZE_CODEC_PM 0x70
+#define AZF_IO_SIZE_CTRL 0x80
+#define AZF_IO_SIZE_CTRL_PM 0x70
-/* the driver initialisation suggests a layout of 4 main areas:
- * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??).
+/* the driver initialisation suggests a layout of 4 areas
+ * within the main card control I/O:
+ * from 0x00 (playback codec), from 0x20 (recording codec)
+ * and from 0x40 (most certainly I2S out codec).
* And another area from 0x60 to 0x6f (DirectX timer, IRQ management,
* power management etc.???). */
-/** playback area **/
-#define IDX_IO_PLAY_FLAGS 0x00 /* PU:0x0000 */
+#define AZF_IO_OFFS_CODEC_PLAYBACK 0x00
+#define AZF_IO_OFFS_CODEC_CAPTURE 0x20
+#define AZF_IO_OFFS_CODEC_I2S_OUT 0x40
+
+#define IDX_IO_CODEC_DMA_FLAGS 0x00 /* PU:0x0000 */
/* able to reactivate output after output muting due to 8/16bit
* output change, just like 0x0002.
* 0x0001 is the only bit that's able to start the DMA counter */
- #define DMA_RESUME 0x0001 /* paused if cleared ? */
+ #define DMA_RESUME 0x0001 /* paused if cleared? */
/* 0x0002 *temporarily* set during DMA stopping. hmm
* both 0x0002 and 0x0004 set in playback setup. */
/* able to reactivate output after output muting due to 8/16bit
* output change, just like 0x0001. */
- #define DMA_PLAY_SOMETHING1 0x0002 /* \ alternated (toggled) */
+ #define DMA_RUN_SOMETHING1 0x0002 /* \ alternated (toggled) */
/* 0x0004: NOT able to reactivate output */
- #define DMA_PLAY_SOMETHING2 0x0004 /* / bits */
+ #define DMA_RUN_SOMETHING2 0x0004 /* / bits */
#define SOMETHING_ALMOST_ALWAYS_SET 0x0008 /* ???; can be modified */
#define DMA_EPILOGUE_SOMETHING 0x0010
#define DMA_SOMETHING_ELSE 0x0020 /* ??? */
- #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused ? not modifiable */
-#define IDX_IO_PLAY_IRQTYPE 0x02 /* PU:0x0001 */
+ #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused? not modifiable */
+#define IDX_IO_CODEC_IRQTYPE 0x02 /* PU:0x0001 */
/* write back to flags in case flags are set, in order to ACK IRQ in handler
* (bit 1 of port 0x64 indicates interrupt for one of these three types)
* sometimes in this case it just writes 0xffff to globally ACK all IRQs
* settings written are not reflected when reading back, though.
- * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows ? */
- #define IRQ_PLAY_SOMETHING 0x0001 /* something & ACK */
- #define IRQ_FINISHED_PLAYBUF_1 0x0002 /* 1st dmabuf finished & ACK */
- #define IRQ_FINISHED_PLAYBUF_2 0x0004 /* 2nd dmabuf finished & ACK */
+ * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows? */
+ #define IRQ_SOMETHING 0x0001 /* something & ACK */
+ #define IRQ_FINISHED_DMABUF_1 0x0002 /* 1st dmabuf finished & ACK */
+ #define IRQ_FINISHED_DMABUF_2 0x0004 /* 2nd dmabuf finished & ACK */
#define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */
#define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */
- #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused ? not modifiable */
-#define IDX_IO_PLAY_DMA_START_1 0x04 /* start address of 1st DMA play area, PU:0x00000000 */
-#define IDX_IO_PLAY_DMA_START_2 0x08 /* start address of 2nd DMA play area, PU:0x00000000 */
-#define IDX_IO_PLAY_DMA_LEN_1 0x0c /* length of 1st DMA play area, PU:0x0000 */
-#define IDX_IO_PLAY_DMA_LEN_2 0x0e /* length of 2nd DMA play area, PU:0x0000 */
-#define IDX_IO_PLAY_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */
-#define IDX_IO_PLAY_DMA_CURROFS 0x14 /* offset within current DMA play area, PU:0x0000 */
-#define IDX_IO_PLAY_SOUNDFORMAT 0x16 /* PU:0x0010 */
+ #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused? not modifiable */
+ /* start address of 1st DMA transfer area, PU:0x00000000 */
+#define IDX_IO_CODEC_DMA_START_1 0x04
+ /* start address of 2nd DMA transfer area, PU:0x00000000 */
+#define IDX_IO_CODEC_DMA_START_2 0x08
+ /* both lengths of DMA transfer areas, PU:0x00000000
+ length1: offset 0x0c, length2: offset 0x0e */
+#define IDX_IO_CODEC_DMA_LENGTHS 0x0c
+#define IDX_IO_CODEC_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */
+ /* offset within current DMA transfer area, PU:0x0000 */
+#define IDX_IO_CODEC_DMA_CURROFS 0x14
+#define IDX_IO_CODEC_SOUNDFORMAT 0x16 /* PU:0x0010 */
/* all unspecified bits can't be modified */
#define SOUNDFORMAT_FREQUENCY_MASK 0x000f
#define SOUNDFORMAT_XTAL1 0x00
@@ -76,6 +85,7 @@
#define SOUNDFORMAT_FLAG_16BIT 0x0010
#define SOUNDFORMAT_FLAG_2CHANNELS 0x0020
+
/* define frequency helpers, for maximum value safety */
enum azf_freq_t {
#define AZF_FREQ(rate) AZF_FREQ_##rate = rate
@@ -96,29 +106,6 @@ enum azf_freq_t {
#undef AZF_FREQ
};
-/** recording area (see also: playback bit flag definitions) **/
-#define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */
-#define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */
- #define IRQ_REC_SOMETHING 0x0001 /* something & ACK */
- #define IRQ_FINISHED_RECBUF_1 0x0002 /* 1st dmabuf finished & ACK */
- #define IRQ_FINISHED_RECBUF_2 0x0004 /* 2nd dmabuf finished & ACK */
- /* hmm, maybe these are just the corresponding *recording* flags ?
- * but OTOH they are most likely at port 0x22 instead */
- #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */
- #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */
-#define IDX_IO_REC_DMA_START_1 0x24 /* PU:0x00000000 */
-#define IDX_IO_REC_DMA_START_2 0x28 /* PU:0x00000000 */
-#define IDX_IO_REC_DMA_LEN_1 0x2c /* PU:0x0000 */
-#define IDX_IO_REC_DMA_LEN_2 0x2e /* PU:0x0000 */
-#define IDX_IO_REC_DMA_CURRPOS 0x30 /* PU:0x00000000 */
-#define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */
-#define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */
-
-/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/
-#define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */
-/* general */
-#define IDX_IO_42H 0x42 /* PU:0x0001 */
-
/** DirectX timer, main interrupt area (FIXME: and something else?) **/
#define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */
/* timer countdown value; triggers IRQ when timer is finished */
@@ -133,17 +120,19 @@ enum azf_freq_t {
#define IDX_IO_IRQSTATUS 0x64
/* some IRQ bit in here might also be used to signal a power-management timer
* timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing).
- * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which
- * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */
+ * OPL3 hardware contains several timers which confusingly in most cases
+ * are NOT routed to an IRQ, but some designs (e.g. LM4560) DO support that,
+ * so I wouldn't be surprised at all to discover that AZF3328
+ * supports that thing as well... */
#define IRQ_PLAYBACK 0x0001
#define IRQ_RECORDING 0x0002
- #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */
+ #define IRQ_I2S_OUT 0x0004 /* this IS I2S, right!? (untested) */
#define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */
#define IRQ_MPU401 0x0010
#define IRQ_TIMER 0x0020 /* DirectX timer */
- #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */
- #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */
+ #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly OPL3 timer? */
+ #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly OPL3 timer? */
#define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */
/* this is set to e.g. 0x3ff or 0x300, and writable;
* maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */
@@ -206,7 +195,7 @@ enum azf_freq_t {
/*** Gameport area port indices ***/
/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */
#define AZF_IO_SIZE_GAME 0x08
-#define AZF_IO_SIZE_GAME_PM 0x06
+#define AZF_IO_SIZE_GAME_PM 0x06
enum {
AZF_GAME_LEGACY_IO_PORT = 0x200
@@ -272,6 +261,12 @@ enum {
* 11 --> 1/200: */
#define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06
+ /* FIXME: these values might be reversed... */
+ #define GAME_HWCFG_ADC_COUNTER_FREQ_STD 0
+ #define GAME_HWCFG_ADC_COUNTER_FREQ_1_2 1
+ #define GAME_HWCFG_ADC_COUNTER_FREQ_1_20 2
+ #define GAME_HWCFG_ADC_COUNTER_FREQ_1_200 3
+
/* enable gameport legacy I/O address (0x200)
* I was unable to locate any configurability for a different address: */
#define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08
@@ -281,6 +276,7 @@ enum {
#define AZF_IO_SIZE_MPU_PM 0x04
/*** OPL3 synth ***/
+/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */
#define AZF_IO_SIZE_OPL3 0x08
#define AZF_IO_SIZE_OPL3_PM 0x06
/* hmm, given that a standard OPL3 has 4 registers only,
@@ -340,4 +336,7 @@ enum {
#define SET_CHAN_LEFT 1
#define SET_CHAN_RIGHT 2
+/* helper macro to align I/O port ranges to 32bit I/O width */
+#define AZF_ALIGN(x) (((x) + 3) & (~3))
+
#endif /* __SOUND_AZT3328_H */
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index 4eb55aa..b518949 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -35,7 +35,7 @@
#ifdef CONFIG_SND_CS46XX_NEW_DSP
-#define CS46XX_MIN_PERIOD_SIZE 1
+#define CS46XX_MIN_PERIOD_SIZE 64
#define CS46XX_MAX_PERIOD_SIZE 1024*1024
#else
#define CS46XX_MIN_PERIOD_SIZE 2048
diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h
index 2d07986..e0394e3 100644
--- a/sound/pci/ctxfi/ct20k2reg.h
+++ b/sound/pci/ctxfi/ct20k2reg.h
@@ -11,9 +11,12 @@
/* Timer Registers */
-#define TIMER_TIMR 0x1B7004
-#define INTERRUPT_GIP 0x1B7010
-#define INTERRUPT_GIE 0x1B7014
+#define WC 0x1b7000
+#define TIMR 0x1b7004
+# define TIMR_IE (1<<15)
+# define TIMR_IP (1<<14)
+#define GIP 0x1b7010
+#define GIE 0x1b7014
/* I2C Registers */
#define I2C_IF_ADDRESS 0x1B9000
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a7f4a67..fee35cf 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -63,7 +63,7 @@ static int amixer_set_input(struct amixer *amixer, struct rsc *rsc)
hw = amixer->rsc.hw;
hw->amixer_set_mode(amixer->rsc.ctrl_blk, AMIXER_Y_IMMEDIATE);
amixer->input = rsc;
- if (NULL == rsc)
+ if (!rsc)
hw->amixer_set_x(amixer->rsc.ctrl_blk, BLANK_SLOT);
else
hw->amixer_set_x(amixer->rsc.ctrl_blk,
@@ -99,7 +99,7 @@ static int amixer_set_sum(struct amixer *amixer, struct sum *sum)
hw = amixer->rsc.hw;
amixer->sum = sum;
- if (NULL == sum) {
+ if (!sum) {
hw->amixer_set_se(amixer->rsc.ctrl_blk, 0);
} else {
hw->amixer_set_se(amixer->rsc.ctrl_blk, 1);
@@ -124,20 +124,20 @@ static int amixer_commit_write(struct amixer *amixer)
/* Program master and conjugate resources */
amixer->rsc.ops->master(&amixer->rsc);
- if (NULL != input)
+ if (input)
input->ops->master(input);
- if (NULL != sum)
+ if (sum)
sum->rsc.ops->master(&sum->rsc);
for (i = 0; i < amixer->rsc.msr; i++) {
hw->amixer_set_dirty_all(amixer->rsc.ctrl_blk);
- if (NULL != input) {
+ if (input) {
hw->amixer_set_x(amixer->rsc.ctrl_blk,
input->ops->output_slot(input));
input->ops->next_conj(input);
}
- if (NULL != sum) {
+ if (sum) {
hw->amixer_set_sadr(amixer->rsc.ctrl_blk,
sum->rsc.ops->index(&sum->rsc));
sum->rsc.ops->next_conj(&sum->rsc);
@@ -147,10 +147,10 @@ static int amixer_commit_write(struct amixer *amixer)
amixer->rsc.ops->next_conj(&amixer->rsc);
}
amixer->rsc.ops->master(&amixer->rsc);
- if (NULL != input)
+ if (input)
input->ops->master(input);
- if (NULL != sum)
+ if (sum)
sum->rsc.ops->master(&sum->rsc);
return 0;
@@ -303,7 +303,7 @@ int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr)
*ramixer_mgr = NULL;
amixer_mgr = kzalloc(sizeof(*amixer_mgr), GFP_KERNEL);
- if (NULL == amixer_mgr)
+ if (!amixer_mgr)
return -ENOMEM;
err = rsc_mgr_init(&amixer_mgr->mgr, AMIXER, AMIXER_RESOURCE_NUM, hw);
@@ -456,7 +456,7 @@ int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr)
*rsum_mgr = NULL;
sum_mgr = kzalloc(sizeof(*sum_mgr), GFP_KERNEL);
- if (NULL == sum_mgr)
+ if (!sum_mgr)
return -ENOMEM;
err = rsc_mgr_init(&sum_mgr->mgr, SUM, SUM_RESOURCE_NUM, hw);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index a49c766..b1b3a64 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -136,7 +136,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
struct snd_pcm_runtime *runtime;
struct ct_vm *vm;
- if (NULL == apcm->substream)
+ if (!apcm->substream)
return 0;
runtime = apcm->substream->runtime;
@@ -144,7 +144,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
apcm->vm_block = vm->map(vm, apcm->substream, runtime->dma_bytes);
- if (NULL == apcm->vm_block)
+ if (!apcm->vm_block)
return -ENOENT;
return 0;
@@ -154,7 +154,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
{
struct ct_vm *vm;
- if (NULL == apcm->vm_block)
+ if (!apcm->vm_block)
return;
vm = atc->vm;
@@ -231,16 +231,16 @@ atc_get_pitch(unsigned int input_rate, unsigned int output_rate)
static int select_rom(unsigned int pitch)
{
- if ((pitch > 0x00428f5c) && (pitch < 0x01b851ec)) {
+ if (pitch > 0x00428f5c && pitch < 0x01b851ec) {
/* 0.26 <= pitch <= 1.72 */
return 1;
- } else if ((0x01d66666 == pitch) || (0x01d66667 == pitch)) {
+ } else if (pitch == 0x01d66666 || pitch == 0x01d66667) {
/* pitch == 1.8375 */
return 2;
- } else if (0x02000000 == pitch) {
+ } else if (pitch == 0x02000000) {
/* pitch == 2 */
return 3;
- } else if ((pitch >= 0x0) && (pitch <= 0x08000000)) {
+ } else if (pitch >= 0x0 && pitch <= 0x08000000) {
/* 0 <= pitch <= 8 */
return 0;
} else {
@@ -283,7 +283,7 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
/* Get AMIXER resource */
n_amixer = (n_amixer < 2) ? 2 : n_amixer;
apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
- if (NULL == apcm->amixers) {
+ if (!apcm->amixers) {
err = -ENOMEM;
goto error1;
}
@@ -311,7 +311,7 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
INIT_VOL, atc->pcm[i+device*2]);
mutex_unlock(&atc->atc_mutex);
src = src->ops->next_interleave(src);
- if (NULL == src)
+ if (!src)
src = apcm->src;
}
@@ -334,7 +334,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
struct srcimp *srcimp;
int i;
- if (NULL != apcm->srcimps) {
+ if (apcm->srcimps) {
for (i = 0; i < apcm->n_srcimp; i++) {
srcimp = apcm->srcimps[i];
srcimp->ops->unmap(srcimp);
@@ -345,7 +345,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
apcm->srcimps = NULL;
}
- if (NULL != apcm->srccs) {
+ if (apcm->srccs) {
for (i = 0; i < apcm->n_srcc; i++) {
src_mgr->put_src(src_mgr, apcm->srccs[i]);
apcm->srccs[i] = NULL;
@@ -354,7 +354,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
apcm->srccs = NULL;
}
- if (NULL != apcm->amixers) {
+ if (apcm->amixers) {
for (i = 0; i < apcm->n_amixer; i++) {
amixer_mgr->put_amixer(amixer_mgr, apcm->amixers[i]);
apcm->amixers[i] = NULL;
@@ -363,17 +363,17 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
apcm->amixers = NULL;
}
- if (NULL != apcm->mono) {
+ if (apcm->mono) {
sum_mgr->put_sum(sum_mgr, apcm->mono);
apcm->mono = NULL;
}
- if (NULL != apcm->src) {
+ if (apcm->src) {
src_mgr->put_src(src_mgr, apcm->src);
apcm->src = NULL;
}
- if (NULL != apcm->vm_block) {
+ if (apcm->vm_block) {
/* Undo device virtual mem map */
ct_unmap_audio_buffer(atc, apcm);
apcm->vm_block = NULL;
@@ -419,7 +419,7 @@ static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm)
src->ops->set_state(src, SRC_STATE_OFF);
src->ops->commit_write(src);
- if (NULL != apcm->srccs) {
+ if (apcm->srccs) {
for (i = 0; i < apcm->n_srcc; i++) {
src = apcm->srccs[i];
src->ops->set_bm(src, 0);
@@ -544,18 +544,18 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
if (n_srcc) {
apcm->srccs = kzalloc(sizeof(void *)*n_srcc, GFP_KERNEL);
- if (NULL == apcm->srccs)
+ if (!apcm->srccs)
return -ENOMEM;
}
if (n_amixer) {
apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
- if (NULL == apcm->amixers) {
+ if (!apcm->amixers) {
err = -ENOMEM;
goto error1;
}
}
apcm->srcimps = kzalloc(sizeof(void *)*n_srcimp, GFP_KERNEL);
- if (NULL == apcm->srcimps) {
+ if (!apcm->srcimps) {
err = -ENOMEM;
goto error1;
}
@@ -818,7 +818,7 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc,
/* Get AMIXER resource */
n_amixer = (n_amixer < 2) ? 2 : n_amixer;
apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
- if (NULL == apcm->amixers) {
+ if (!apcm->amixers) {
err = -ENOMEM;
goto error1;
}
@@ -919,7 +919,7 @@ spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
amixer = apcm->amixers[i];
amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL);
src = src->ops->next_interleave(src);
- if (NULL == src)
+ if (!src)
src = apcm->src;
}
/* Connect to SPDIFOO */
@@ -1121,7 +1121,7 @@ static int atc_release_resources(struct ct_atc *atc)
struct ct_mixer *mixer = NULL;
/* disconnect internal mixer objects */
- if (NULL != atc->mixer) {
+ if (atc->mixer) {
mixer = atc->mixer;
mixer->set_input_left(mixer, MIX_LINE_IN, NULL);
mixer->set_input_right(mixer, MIX_LINE_IN, NULL);
@@ -1131,7 +1131,7 @@ static int atc_release_resources(struct ct_atc *atc)
mixer->set_input_right(mixer, MIX_SPDIF_IN, NULL);
}
- if (NULL != atc->daios) {
+ if (atc->daios) {
daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
for (i = 0; i < atc->n_daio; i++) {
daio = atc->daios[i];
@@ -1149,7 +1149,7 @@ static int atc_release_resources(struct ct_atc *atc)
atc->daios = NULL;
}
- if (NULL != atc->pcm) {
+ if (atc->pcm) {
sum_mgr = atc->rsc_mgrs[SUM];
for (i = 0; i < atc->n_pcm; i++)
sum_mgr->put_sum(sum_mgr, atc->pcm[i]);
@@ -1158,7 +1158,7 @@ static int atc_release_resources(struct ct_atc *atc)
atc->pcm = NULL;
}
- if (NULL != atc->srcs) {
+ if (atc->srcs) {
src_mgr = atc->rsc_mgrs[SRC];
for (i = 0; i < atc->n_src; i++)
src_mgr->put_src(src_mgr, atc->srcs[i]);
@@ -1167,7 +1167,7 @@ static int atc_release_resources(struct ct_atc *atc)
atc->srcs = NULL;
}
- if (NULL != atc->srcimps) {
+ if (atc->srcimps) {
srcimp_mgr = atc->rsc_mgrs[SRCIMP];
for (i = 0; i < atc->n_srcimp; i++) {
srcimp = atc->srcimps[i];
@@ -1185,7 +1185,7 @@ static int ct_atc_destroy(struct ct_atc *atc)
{
int i = 0;
- if (NULL == atc)
+ if (!atc)
return 0;
if (atc->timer) {
@@ -1196,21 +1196,20 @@ static int ct_atc_destroy(struct ct_atc *atc)
atc_release_resources(atc);
/* Destroy internal mixer objects */
- if (NULL != atc->mixer)
+ if (atc->mixer)
ct_mixer_destroy(atc->mixer);
for (i = 0; i < NUM_RSCTYP; i++) {
- if ((NULL != rsc_mgr_funcs[i].destroy) &&
- (NULL != atc->rsc_mgrs[i]))
+ if (rsc_mgr_funcs[i].destroy && atc->rsc_mgrs[i])
rsc_mgr_funcs[i].destroy(atc->rsc_mgrs[i]);
}
- if (NULL != atc->hw)
+ if (atc->hw)
destroy_hw_obj((struct hw *)atc->hw);
/* Destroy device virtual memory manager object */
- if (NULL != atc->vm) {
+ if (atc->vm) {
ct_vm_destroy(atc->vm);
atc->vm = NULL;
}
@@ -1275,7 +1274,7 @@ int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc)
alsa_dev_funcs[MIXER].public_name = atc->chip_name;
for (i = 0; i < NUM_CTALSADEVS; i++) {
- if (NULL == alsa_dev_funcs[i].create)
+ if (!alsa_dev_funcs[i].create)
continue;
err = alsa_dev_funcs[i].create(atc, i,
@@ -1312,7 +1311,7 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc)
return err;
for (i = 0; i < NUM_RSCTYP; i++) {
- if (NULL == rsc_mgr_funcs[i].create)
+ if (!rsc_mgr_funcs[i].create)
continue;
err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]);
@@ -1339,19 +1338,19 @@ static int atc_get_resources(struct ct_atc *atc)
int err, i;
atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL);
- if (NULL == atc->daios)
+ if (!atc->daios)
return -ENOMEM;
atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
- if (NULL == atc->srcs)
+ if (!atc->srcs)
return -ENOMEM;
atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
- if (NULL == atc->srcimps)
+ if (!atc->srcimps)
return -ENOMEM;
atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL);
- if (NULL == atc->pcm)
+ if (!atc->pcm)
return -ENOMEM;
daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
@@ -1648,7 +1647,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
*ratc = NULL;
atc = kzalloc(sizeof(*atc), GFP_KERNEL);
- if (NULL == atc)
+ if (!atc)
return -ENOMEM;
/* Set operations */
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index deb6cfa..af56eb9 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -173,7 +173,7 @@ static int dao_set_left_input(struct dao *dao, struct rsc *input)
int i;
entry = kzalloc((sizeof(*entry) * daio->rscl.msr), GFP_KERNEL);
- if (NULL == entry)
+ if (!entry)
return -ENOMEM;
/* Program master and conjugate resources */
@@ -201,7 +201,7 @@ static int dao_set_right_input(struct dao *dao, struct rsc *input)
int i;
entry = kzalloc((sizeof(*entry) * daio->rscr.msr), GFP_KERNEL);
- if (NULL == entry)
+ if (!entry)
return -ENOMEM;
/* Program master and conjugate resources */
@@ -228,7 +228,7 @@ static int dao_clear_left_input(struct dao *dao)
struct daio *daio = &dao->daio;
int i;
- if (NULL == dao->imappers[0])
+ if (!dao->imappers[0])
return 0;
entry = dao->imappers[0];
@@ -252,7 +252,7 @@ static int dao_clear_right_input(struct dao *dao)
struct daio *daio = &dao->daio;
int i;
- if (NULL == dao->imappers[daio->rscl.msr])
+ if (!dao->imappers[daio->rscl.msr])
return 0;
entry = dao->imappers[daio->rscl.msr];
@@ -408,7 +408,7 @@ static int dao_rsc_init(struct dao *dao,
return err;
dao->imappers = kzalloc(sizeof(void *)*desc->msr*2, GFP_KERNEL);
- if (NULL == dao->imappers) {
+ if (!dao->imappers) {
err = -ENOMEM;
goto error1;
}
@@ -442,11 +442,11 @@ error1:
static int dao_rsc_uninit(struct dao *dao)
{
- if (NULL != dao->imappers) {
- if (NULL != dao->imappers[0])
+ if (dao->imappers) {
+ if (dao->imappers[0])
dao_clear_left_input(dao);
- if (NULL != dao->imappers[dao->daio.rscl.msr])
+ if (dao->imappers[dao->daio.rscl.msr])
dao_clear_right_input(dao);
kfree(dao->imappers);
@@ -555,7 +555,7 @@ static int get_daio_rsc(struct daio_mgr *mgr,
/* Allocate mem for daio resource */
if (desc->type <= DAIO_OUT_MAX) {
dao = kzalloc(sizeof(*dao), GFP_KERNEL);
- if (NULL == dao) {
+ if (!dao) {
err = -ENOMEM;
goto error;
}
@@ -566,7 +566,7 @@ static int get_daio_rsc(struct daio_mgr *mgr,
*rdaio = &dao->daio;
} else {
dai = kzalloc(sizeof(*dai), GFP_KERNEL);
- if (NULL == dai) {
+ if (!dai) {
err = -ENOMEM;
goto error;
}
@@ -583,9 +583,9 @@ static int get_daio_rsc(struct daio_mgr *mgr,
return 0;
error:
- if (NULL != dao)
+ if (dao)
kfree(dao);
- else if (NULL != dai)
+ else if (dai)
kfree(dai);
spin_lock_irqsave(&mgr->mgr_lock, flags);
@@ -663,7 +663,7 @@ static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry)
int err;
spin_lock_irqsave(&mgr->imap_lock, flags);
- if ((0 == entry->addr) && (mgr->init_imap_added)) {
+ if (!entry->addr && mgr->init_imap_added) {
input_mapper_delete(&mgr->imappers, mgr->init_imap,
daio_map_op, mgr);
mgr->init_imap_added = 0;
@@ -707,7 +707,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr)
*rdaio_mgr = NULL;
daio_mgr = kzalloc(sizeof(*daio_mgr), GFP_KERNEL);
- if (NULL == daio_mgr)
+ if (!daio_mgr)
return -ENOMEM;
err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw);
@@ -718,7 +718,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr)
spin_lock_init(&daio_mgr->imap_lock);
INIT_LIST_HEAD(&daio_mgr->imappers);
entry = kzalloc(sizeof(*entry), GFP_KERNEL);
- if (NULL == entry) {
+ if (!entry) {
err = -ENOMEM;
goto error2;
}
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index ad3e1d1..0cf400f 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -168,7 +168,7 @@ static int src_get_rsc_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -494,7 +494,7 @@ static int src_mgr_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -515,7 +515,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -723,7 +723,7 @@ static int amixer_mgr_get_ctrl_blk(void **rblk)
*rblk = NULL;
/*blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;*/
@@ -909,7 +909,7 @@ static int dai_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -958,7 +958,7 @@ static int dao_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -1152,7 +1152,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
blk->i2sctl = hw_read_20kx(hw, I2SCTL);
@@ -1808,7 +1808,7 @@ static int uaa_to_xfi(struct pci_dev *pci)
/* By default, Hendrix card UAA Bar0 should be using memory... */
io_base = pci_resource_start(pci, 0);
mem_base = ioremap(io_base, pci_resource_len(pci, 0));
- if (NULL == mem_base)
+ if (!mem_base)
return -ENOENT;
/* Read current mode from Mode Change Register */
@@ -1977,7 +1977,7 @@ static int hw_card_shutdown(struct hw *hw)
hw->irq = -1;
- if (NULL != ((void *)hw->mem_base))
+ if (hw->mem_base)
iounmap((void *)hw->mem_base);
hw->mem_base = (unsigned long)NULL;
@@ -2274,7 +2274,7 @@ int __devinit create_20k1_hw_obj(struct hw **rhw)
*rhw = NULL;
hw20k1 = kzalloc(sizeof(*hw20k1), GFP_KERNEL);
- if (NULL == hw20k1)
+ if (!hw20k1)
return -ENOMEM;
spin_lock_init(&hw20k1->reg_20k1_lock);
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index dec46d0..b6b11bf 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -166,7 +166,7 @@ static int src_get_rsc_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -492,7 +492,7 @@ static int src_mgr_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -513,7 +513,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -891,7 +891,7 @@ static int dai_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -941,7 +941,7 @@ static int dao_get_ctrl_blk(void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
*rblk = blk;
@@ -1092,7 +1092,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk)
*rblk = NULL;
blk = kzalloc(sizeof(*blk), GFP_KERNEL);
- if (NULL == blk)
+ if (!blk)
return -ENOMEM;
for (i = 0; i < 8; i++) {
@@ -1112,6 +1112,26 @@ static int daio_mgr_put_ctrl_blk(void *blk)
return 0;
}
+/* Timer interrupt */
+static int set_timer_irq(struct hw *hw, int enable)
+{
+ hw_write_20kx(hw, GIE, enable ? IT_INT : 0);
+ return 0;
+}
+
+static int set_timer_tick(struct hw *hw, unsigned int ticks)
+{
+ if (ticks)
+ ticks |= TIMR_IE | TIMR_IP;
+ hw_write_20kx(hw, TIMR, ticks);
+ return 0;
+}
+
+static unsigned int get_wc(struct hw *hw)
+{
+ return hw_read_20kx(hw, WC);
+}
+
/* Card hardware initialization block */
struct dac_conf {
unsigned int msr; /* master sample rate in rsrs */
@@ -1841,6 +1861,22 @@ static int hw_have_digit_io_switch(struct hw *hw)
return 0;
}
+static irqreturn_t ct_20k2_interrupt(int irq, void *dev_id)
+{
+ struct hw *hw = dev_id;
+ unsigned int status;
+
+ status = hw_read_20kx(hw, GIP);
+ if (!status)
+ return IRQ_NONE;
+
+ if (hw->irq_callback)
+ hw->irq_callback(hw->irq_callback_data, status);
+
+ hw_write_20kx(hw, GIP, status);
+ return IRQ_HANDLED;
+}
+
static int hw_card_start(struct hw *hw)
{
int err = 0;
@@ -1868,7 +1904,7 @@ static int hw_card_start(struct hw *hw)
hw->io_base = pci_resource_start(hw->pci, 2);
hw->mem_base = (unsigned long)ioremap(hw->io_base,
pci_resource_len(hw->pci, 2));
- if (NULL == (void *)hw->mem_base) {
+ if (!hw->mem_base) {
err = -ENOENT;
goto error2;
}
@@ -1879,12 +1915,15 @@ static int hw_card_start(struct hw *hw)
set_field(&gctl, GCTL_UAA, 0);
hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
- /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED,
- atc->chip_details->nm_card, hw))) {
- goto error3;
+ if (hw->irq < 0) {
+ err = request_irq(pci->irq, ct_20k2_interrupt, IRQF_SHARED,
+ "ctxfi", hw);
+ if (err < 0) {
+ printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
+ goto error2;
+ }
+ hw->irq = pci->irq;
}
- hw->irq = pci->irq;
- */
pci_set_master(pci);
@@ -1923,7 +1962,7 @@ static int hw_card_shutdown(struct hw *hw)
hw->irq = -1;
- if (NULL != ((void *)hw->mem_base))
+ if (hw->mem_base)
iounmap((void *)hw->mem_base);
hw->mem_base = (unsigned long)NULL;
@@ -1972,7 +2011,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
/* Reset all global pending interrupts */
- hw_write_20kx(hw, INTERRUPT_GIE, 0);
+ hw_write_20kx(hw, GIE, 0);
/* Reset all SRC pending interrupts */
hw_write_20kx(hw, SRC_IP, 0);
@@ -2149,6 +2188,10 @@ static struct hw ct20k2_preset __devinitdata = {
.daio_mgr_set_imapnxt = daio_mgr_set_imapnxt,
.daio_mgr_set_imapaddr = daio_mgr_set_imapaddr,
.daio_mgr_commit_write = daio_mgr_commit_write,
+
+ .set_timer_irq = set_timer_irq,
+ .set_timer_tick = set_timer_tick,
+ .get_wc = get_wc,
};
int __devinit create_20k2_hw_obj(struct hw **rhw)
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
index f26d7cd..15c1e72 100644
--- a/sound/pci/ctxfi/ctmixer.c
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -654,7 +654,7 @@ ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new)
int err;
kctl = snd_ctl_new1(new, mixer->atc);
- if (NULL == kctl)
+ if (!kctl)
return -ENOMEM;
if (SNDRV_CTL_ELEM_IFACE_PCM == kctl->id.iface)
@@ -837,17 +837,17 @@ static int ct_mixer_get_mem(struct ct_mixer **rmixer)
*rmixer = NULL;
/* Allocate mem for mixer obj */
mixer = kzalloc(sizeof(*mixer), GFP_KERNEL);
- if (NULL == mixer)
+ if (!mixer)
return -ENOMEM;
mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM),
GFP_KERNEL);
- if (NULL == mixer->amixers) {
+ if (!mixer->amixers) {
err = -ENOMEM;
goto error1;
}
mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL);
- if (NULL == mixer->sums) {
+ if (!mixer->sums) {
err = -ENOMEM;
goto error2;
}
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 60ea231..d0dc227 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -97,7 +97,7 @@ static void ct_atc_pcm_interrupt(struct ct_atc_pcm *atc_pcm)
{
struct ct_atc_pcm *apcm = atc_pcm;
- if (NULL == apcm->substream)
+ if (!apcm->substream)
return;
snd_pcm_period_elapsed(apcm->substream);
@@ -123,7 +123,7 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
int err;
apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
- if (NULL == apcm)
+ if (!apcm)
return -ENOMEM;
apcm->substream = substream;
@@ -271,7 +271,7 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
int err;
apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
- if (NULL == apcm)
+ if (!apcm)
return -ENOMEM;
apcm->started = 0;
diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c
index 889c495..7dfaf67 100644
--- a/sound/pci/ctxfi/ctresource.c
+++ b/sound/pci/ctxfi/ctresource.c
@@ -144,7 +144,7 @@ int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw)
rsc->msr = msr;
rsc->hw = hw;
rsc->ops = &rsc_generic_ops;
- if (NULL == hw) {
+ if (!hw) {
rsc->ctrl_blk = NULL;
return 0;
}
@@ -216,7 +216,7 @@ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type,
mgr->type = NUM_RSCTYP;
mgr->rscs = kzalloc(((amount + 8 - 1) / 8), GFP_KERNEL);
- if (NULL == mgr->rscs)
+ if (!mgr->rscs)
return -ENOMEM;
switch (type) {
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index df43a5c..c749fa7 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -441,7 +441,7 @@ get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc)
else
src = kzalloc(sizeof(*src), GFP_KERNEL);
- if (NULL == src) {
+ if (!src) {
err = -ENOMEM;
goto error1;
}
@@ -550,7 +550,7 @@ int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr)
*rsrc_mgr = NULL;
src_mgr = kzalloc(sizeof(*src_mgr), GFP_KERNEL);
- if (NULL == src_mgr)
+ if (!src_mgr)
return -ENOMEM;
err = rsc_mgr_init(&src_mgr->mgr, SRC, SRC_RESOURCE_NUM, hw);
@@ -679,7 +679,7 @@ static int srcimp_rsc_init(struct srcimp *srcimp,
/* Reserve memory for imapper nodes */
srcimp->imappers = kzalloc(sizeof(struct imapper)*desc->msr,
GFP_KERNEL);
- if (NULL == srcimp->imappers) {
+ if (!srcimp->imappers) {
err = -ENOMEM;
goto error1;
}
@@ -833,7 +833,7 @@ int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr)
*rsrcimp_mgr = NULL;
srcimp_mgr = kzalloc(sizeof(*srcimp_mgr), GFP_KERNEL);
- if (NULL == srcimp_mgr)
+ if (!srcimp_mgr)
return -ENOMEM;
err = rsc_mgr_init(&srcimp_mgr->mgr, SRCIMP, SRCIMP_RESOURCE_NUM, hw);
@@ -844,7 +844,7 @@ int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr)
spin_lock_init(&srcimp_mgr->imap_lock);
INIT_LIST_HEAD(&srcimp_mgr->imappers);
entry = kzalloc(sizeof(*entry), GFP_KERNEL);
- if (NULL == entry) {
+ if (!entry) {
err = -ENOMEM;
goto error2;
}
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
index 67665a7..6b78752 100644
--- a/sound/pci/ctxfi/ctvmem.c
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -60,7 +60,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size)
}
block = kzalloc(sizeof(*block), GFP_KERNEL);
- if (NULL == block)
+ if (!block)
goto out;
block->addr = entry->addr;
@@ -181,7 +181,7 @@ int ct_vm_create(struct ct_vm **rvm)
*rvm = NULL;
vm = kzalloc(sizeof(*vm), GFP_KERNEL);
- if (NULL == vm)
+ if (!vm)
return -ENOMEM;
mutex_init(&vm->lock);
@@ -189,7 +189,7 @@ int ct_vm_create(struct ct_vm **rvm)
/* Allocate page table pages */
for (i = 0; i < CT_PTP_NUM; i++) {
vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL);
- if (NULL == vm->ptp[i])
+ if (!vm->ptp[i])
break;
}
if (!i) {
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 04438f1..55545e0 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -46,6 +46,20 @@ config SND_HDA_INPUT_JACK
Say Y here to enable the jack plugging notification via
input layer.
+config SND_HDA_PATCH_LOADER
+ bool "Support initialization patch loading for HD-audio"
+ depends on EXPERIMENTAL
+ select FW_LOADER
+ select SND_HDA_HWDEP
+ select SND_HDA_RECONFIG
+ help
+ Say Y here to allow the HD-audio driver to load a pseudo
+ firmware file ("patch") for overriding the BIOS setup at
+ start up. The "patch" file can be specified via patch module
+ option, such as patch=hda-init.
+
+ This option turns on hwdep and reconfig features automatically.
+
config SND_HDA_CODEC_REALTEK
bool "Build Realtek HD-audio codec support"
default y
@@ -134,6 +148,19 @@ config SND_HDA_ELD
def_bool y
depends on SND_HDA_CODEC_INTELHDMI
+config SND_HDA_CODEC_CIRRUS
+ bool "Build Cirrus Logic codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Cirrus Logic codec support in
+ snd-hda-intel driver, such as CS4206.
+
+ When the HD-audio driver is built as a module, the codec
+ support code is also built as another module,
+ snd-hda-codec-cirrus.
+ This module is automatically loaded at probing.
+
config SND_HDA_CODEC_CONEXANT
bool "Build Conexant HD-audio codec support"
default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index e3081d4..315a1c4 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@ snd-hda-codec-analog-objs := patch_analog.o
snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-atihdmi-objs := patch_atihdmi.o
+snd-hda-codec-cirrus-objs := patch_cirrus.o
snd-hda-codec-ca0110-objs := patch_ca0110.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
@@ -41,6 +42,9 @@ endif
ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
endif
+ifdef CONFIG_SND_HDA_CODEC_CIRRUS
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o
+endif
ifdef CONFIG_SND_HDA_CODEC_CA0110
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
endif
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index b0275a0..3f51a98 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -24,6 +24,7 @@
#include <linux/workqueue.h>
#include <sound/core.h>
#include "hda_beep.h"
+#include "hda_local.h"
enum {
DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */
@@ -118,6 +119,9 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
struct hda_beep *beep;
int err;
+ if (!snd_hda_get_bool_hint(codec, "beep"))
+ return 0; /* disabled explicitly */
+
beep = kzalloc(sizeof(*beep), GFP_KERNEL);
if (beep == NULL)
return -ENOMEM;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c7df01b..af989f6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -44,6 +44,7 @@ struct hda_vendor_id {
/* codec vendor labels */
static struct hda_vendor_id hda_vendor_ids[] = {
{ 0x1002, "ATI" },
+ { 0x1013, "Cirrus Logic" },
{ 0x1057, "Motorola" },
{ 0x1095, "Silicon Image" },
{ 0x10de, "Nvidia" },
@@ -150,7 +151,14 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
{
u32 val;
- val = (u32)(codec->addr & 0x0f) << 28;
+ if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) ||
+ (verb & ~0xfff) || (parm & ~0xffff)) {
+ printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n",
+ codec->addr, direct, nid, verb, parm);
+ return ~0;
+ }
+
+ val = (u32)codec->addr << 28;
val |= (u32)direct << 27;
val |= (u32)nid << 20;
val |= verb << 8;
@@ -167,6 +175,9 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
struct hda_bus *bus = codec->bus;
int err;
+ if (cmd == ~0)
+ return -1;
+
if (res)
*res = -1;
again:
@@ -291,11 +302,20 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
unsigned int parm;
int i, conn_len, conns;
unsigned int shift, num_elems, mask;
+ unsigned int wcaps;
hda_nid_t prev_nid;
if (snd_BUG_ON(!conn_list || max_conns <= 0))
return -EINVAL;
+ wcaps = get_wcaps(codec, nid);
+ if (!(wcaps & AC_WCAP_CONN_LIST) &&
+ get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
+ snd_printk(KERN_WARNING "hda_codec: "
+ "connection list not available for 0x%x\n", nid);
+ return -EINVAL;
+ }
+
parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
if (parm & AC_CLIST_LONG) {
/* long form */
@@ -316,6 +336,8 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
/* single connection */
parm = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CONNECT_LIST, 0);
+ if (parm == -1 && codec->bus->rirb_error)
+ return -EIO;
conn_list[0] = parm & mask;
return 1;
}
@@ -327,9 +349,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
int range_val;
hda_nid_t val, n;
- if (i % num_elems == 0)
+ if (i % num_elems == 0) {
parm = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CONNECT_LIST, i);
+ if (parm == -1 && codec->bus->rirb_error)
+ return -EIO;
+ }
range_val = !!(parm & (1 << (shift-1))); /* ranges */
val = parm & mask;
if (val == 0) {
@@ -727,8 +752,7 @@ static int read_pin_defaults(struct hda_codec *codec)
for (i = 0; i < codec->num_nodes; i++, nid++) {
struct hda_pincfg *pin;
unsigned int wcaps = get_wcaps(codec, nid);
- unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
- AC_WCAP_TYPE_SHIFT;
+ unsigned int wid_type = get_wcaps_type(wcaps);
if (wid_type != AC_WID_PIN)
continue;
pin = snd_array_new(&codec->init_pins);
@@ -891,7 +915,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
* Returns 0 if successful, or a negative error code.
*/
int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
- int do_init, struct hda_codec **codecp)
+ struct hda_codec **codecp)
{
struct hda_codec *codec;
char component[31];
@@ -984,11 +1008,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D0);
- if (do_init) {
- err = snd_hda_codec_configure(codec);
- if (err < 0)
- goto error;
- }
snd_hda_codec_proc_new(codec);
snd_hda_create_hwdep(codec);
@@ -1042,6 +1061,7 @@ int snd_hda_codec_configure(struct hda_codec *codec)
err = init_unsol_queue(codec->bus);
return err;
}
+EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
/**
* snd_hda_codec_setup_stream - set up the codec for streaming
@@ -2356,16 +2376,20 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
hda_nid_t nid;
int i;
- snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+ /* this delay seems necessary to avoid click noise at power-down */
+ if (power_state == AC_PWRST_D3)
+ msleep(100);
+ snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
power_state);
- msleep(10); /* partial workaround for "azx_get_response timeout" */
+ /* partial workaround for "azx_get_response timeout" */
+ if (power_state == AC_PWRST_D0)
+ msleep(10);
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
unsigned int wcaps = get_wcaps(codec, nid);
if (wcaps & AC_WCAP_POWER) {
- unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
- AC_WCAP_TYPE_SHIFT;
+ unsigned int wid_type = get_wcaps_type(wcaps);
if (power_state == AC_PWRST_D3 &&
wid_type == AC_WID_PIN) {
unsigned int pincap;
@@ -2573,7 +2597,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
case 20:
case 24:
case 32:
- if (maxbps >= 32)
+ if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE)
val |= 0x40;
else if (maxbps >= 24)
val |= 0x30;
@@ -2700,11 +2724,12 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
bps = 20;
}
}
- else if (streams == AC_SUPFMT_FLOAT32) {
- /* should be exclusive */
+ if (streams & AC_SUPFMT_FLOAT32) {
formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
- bps = 32;
- } else if (streams == AC_SUPFMT_AC3) {
+ if (!bps)
+ bps = 32;
+ }
+ if (streams == AC_SUPFMT_AC3) {
/* should be exclusive */
/* temporary hack: we have still no proper support
* for the direct AC3 stream...
@@ -3102,7 +3127,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
tbl = q;
if (tbl->value >= 0 && tbl->value < num_configs) {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
char tmp[10];
const char *model = NULL;
if (models)
@@ -3655,8 +3680,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
end_nid = codec->start_nid + codec->num_nodes;
for (nid = codec->start_nid; nid < end_nid; nid++) {
unsigned int wid_caps = get_wcaps(codec, nid);
- unsigned int wid_type =
- (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ unsigned int wid_type = get_wcaps_type(wid_caps);
unsigned int def_conf;
short assoc, loc;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 1b75f28..99552fb 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -830,7 +830,8 @@ enum {
int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
struct hda_bus **busp);
int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
- int do_init, struct hda_codec **codecp);
+ struct hda_codec **codecp);
+int snd_hda_codec_configure(struct hda_codec *codec);
/*
* low level functions
@@ -938,6 +939,13 @@ static inline void snd_hda_power_down(struct hda_codec *codec) {}
#define snd_hda_codec_needs_resume(codec) 1
#endif
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+/*
+ * patch firmware
+ */
+int snd_hda_load_patch(struct hda_bus *bus, const char *patch);
+#endif
+
/*
* Codec modularization
*/
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 1d5797a..b36f6c5 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -121,11 +121,17 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
if (node == NULL)
return -ENOMEM;
node->nid = nid;
- nconns = snd_hda_get_connections(codec, nid, conn_list,
- HDA_MAX_CONNECTIONS);
- if (nconns < 0) {
- kfree(node);
- return nconns;
+ node->wid_caps = get_wcaps(codec, nid);
+ node->type = get_wcaps_type(node->wid_caps);
+ if (node->wid_caps & AC_WCAP_CONN_LIST) {
+ nconns = snd_hda_get_connections(codec, nid, conn_list,
+ HDA_MAX_CONNECTIONS);
+ if (nconns < 0) {
+ kfree(node);
+ return nconns;
+ }
+ } else {
+ nconns = 0;
}
if (nconns <= ARRAY_SIZE(node->slist))
node->conn_list = node->slist;
@@ -140,8 +146,6 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
}
memcpy(node->conn_list, conn_list, nconns * sizeof(hda_nid_t));
node->nconns = nconns;
- node->wid_caps = get_wcaps(codec, nid);
- node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
if (node->type == AC_WID_PIN) {
node->pin_caps = snd_hda_query_pin_caps(codec, node->nid);
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 6812fbe..cc24e67 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -24,6 +24,7 @@
#include <linux/compat.h>
#include <linux/mutex.h>
#include <linux/ctype.h>
+#include <linux/firmware.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
@@ -312,12 +313,8 @@ static ssize_t init_verbs_show(struct device *dev,
return len;
}
-static ssize_t init_verbs_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf, size_t count)
+static int parse_init_verbs(struct hda_codec *codec, const char *buf)
{
- struct snd_hwdep *hwdep = dev_get_drvdata(dev);
- struct hda_codec *codec = hwdep->private_data;
struct hda_verb *v;
int nid, verb, param;
@@ -331,6 +328,18 @@ static ssize_t init_verbs_store(struct device *dev,
v->nid = nid;
v->verb = verb;
v->param = param;
+ return 0;
+}
+
+static ssize_t init_verbs_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+ struct hda_codec *codec = hwdep->private_data;
+ int err = parse_init_verbs(codec, buf);
+ if (err < 0)
+ return err;
return count;
}
@@ -376,19 +385,15 @@ static void remove_trail_spaces(char *str)
#define MAX_HINTS 1024
-static ssize_t hints_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf, size_t count)
+static int parse_hints(struct hda_codec *codec, const char *buf)
{
- struct snd_hwdep *hwdep = dev_get_drvdata(dev);
- struct hda_codec *codec = hwdep->private_data;
char *key, *val;
struct hda_hint *hint;
while (isspace(*buf))
buf++;
if (!*buf || *buf == '#' || *buf == '\n')
- return count;
+ return 0;
if (*buf == '=')
return -EINVAL;
key = kstrndup_noeol(buf, 1024);
@@ -411,7 +416,7 @@ static ssize_t hints_store(struct device *dev,
kfree(hint->key);
hint->key = key;
hint->val = val;
- return count;
+ return 0;
}
/* allocate a new hint entry */
if (codec->hints.used >= MAX_HINTS)
@@ -424,6 +429,18 @@ static ssize_t hints_store(struct device *dev,
}
hint->key = key;
hint->val = val;
+ return 0;
+}
+
+static ssize_t hints_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+ struct hda_codec *codec = hwdep->private_data;
+ int err = parse_hints(codec, buf);
+ if (err < 0)
+ return err;
return count;
}
@@ -469,20 +486,24 @@ static ssize_t driver_pin_configs_show(struct device *dev,
#define MAX_PIN_CONFIGS 32
-static ssize_t user_pin_configs_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf, size_t count)
+static int parse_user_pin_configs(struct hda_codec *codec, const char *buf)
{
- struct snd_hwdep *hwdep = dev_get_drvdata(dev);
- struct hda_codec *codec = hwdep->private_data;
int nid, cfg;
- int err;
if (sscanf(buf, "%i %i", &nid, &cfg) != 2)
return -EINVAL;
if (!nid)
return -EINVAL;
- err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+ return snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+}
+
+static ssize_t user_pin_configs_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+ struct hda_codec *codec = hwdep->private_data;
+ int err = parse_user_pin_configs(codec, buf);
if (err < 0)
return err;
return count;
@@ -553,3 +574,180 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
#endif /* CONFIG_SND_HDA_RECONFIG */
+
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+
+/* parser mode */
+enum {
+ LINE_MODE_NONE,
+ LINE_MODE_CODEC,
+ LINE_MODE_MODEL,
+ LINE_MODE_PINCFG,
+ LINE_MODE_VERB,
+ LINE_MODE_HINT,
+ NUM_LINE_MODES,
+};
+
+static inline int strmatch(const char *a, const char *b)
+{
+ return strnicmp(a, b, strlen(b)) == 0;
+}
+
+/* parse the contents after the line "[codec]"
+ * accept only the line with three numbers, and assign the current codec
+ */
+static void parse_codec_mode(char *buf, struct hda_bus *bus,
+ struct hda_codec **codecp)
+{
+ unsigned int vendorid, subid, caddr;
+ struct hda_codec *codec;
+
+ *codecp = NULL;
+ if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) {
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ if (codec->addr == caddr) {
+ *codecp = codec;
+ break;
+ }
+ }
+ }
+}
+
+/* parse the contents after the other command tags, [pincfg], [verb],
+ * [hint] and [model]
+ * just pass to the sysfs helper (only when any codec was specified)
+ */
+static void parse_pincfg_mode(char *buf, struct hda_bus *bus,
+ struct hda_codec **codecp)
+{
+ if (!*codecp)
+ return;
+ parse_user_pin_configs(*codecp, buf);
+}
+
+static void parse_verb_mode(char *buf, struct hda_bus *bus,
+ struct hda_codec **codecp)
+{
+ if (!*codecp)
+ return;
+ parse_init_verbs(*codecp, buf);
+}
+
+static void parse_hint_mode(char *buf, struct hda_bus *bus,
+ struct hda_codec **codecp)
+{
+ if (!*codecp)
+ return;
+ parse_hints(*codecp, buf);
+}
+
+static void parse_model_mode(char *buf, struct hda_bus *bus,
+ struct hda_codec **codecp)
+{
+ if (!*codecp)
+ return;
+ kfree((*codecp)->modelname);
+ (*codecp)->modelname = kstrdup(buf, GFP_KERNEL);
+}
+
+struct hda_patch_item {
+ const char *tag;
+ void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc);
+};
+
+static struct hda_patch_item patch_items[NUM_LINE_MODES] = {
+ [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode },
+ [LINE_MODE_MODEL] = { "[model]", parse_model_mode },
+ [LINE_MODE_VERB] = { "[verb]", parse_verb_mode },
+ [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode },
+ [LINE_MODE_HINT] = { "[hint]", parse_hint_mode },
+};
+
+/* check the line starting with '[' -- change the parser mode accodingly */
+static int parse_line_mode(char *buf, struct hda_bus *bus)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(patch_items); i++) {
+ if (!patch_items[i].tag)
+ continue;
+ if (strmatch(buf, patch_items[i].tag))
+ return i;
+ }
+ return LINE_MODE_NONE;
+}
+
+/* copy one line from the buffer in fw, and update the fields in fw
+ * return zero if it reaches to the end of the buffer, or non-zero
+ * if successfully copied a line
+ *
+ * the spaces at the beginning and the end of the line are stripped
+ */
+static int get_line_from_fw(char *buf, int size, struct firmware *fw)
+{
+ int len;
+ const char *p = fw->data;
+ while (isspace(*p) && fw->size) {
+ p++;
+ fw->size--;
+ }
+ if (!fw->size)
+ return 0;
+ if (size < fw->size)
+ size = fw->size;
+
+ for (len = 0; len < fw->size; len++) {
+ if (!*p)
+ break;
+ if (*p == '\n') {
+ p++;
+ len++;
+ break;
+ }
+ if (len < size)
+ *buf++ = *p++;
+ }
+ *buf = 0;
+ fw->size -= len;
+ fw->data = p;
+ remove_trail_spaces(buf);
+ return 1;
+}
+
+/*
+ * load a "patch" firmware file and parse it
+ */
+int snd_hda_load_patch(struct hda_bus *bus, const char *patch)
+{
+ int err;
+ const struct firmware *fw;
+ struct firmware tmp;
+ char buf[128];
+ struct hda_codec *codec;
+ int line_mode;
+ struct device *dev = bus->card->dev;
+
+ if (snd_BUG_ON(!dev))
+ return -ENODEV;
+ err = request_firmware(&fw, patch, dev);
+ if (err < 0) {
+ printk(KERN_ERR "hda-codec: Cannot load the patch '%s'\n",
+ patch);
+ return err;
+ }
+
+ tmp = *fw;
+ line_mode = LINE_MODE_NONE;
+ codec = NULL;
+ while (get_line_from_fw(buf, sizeof(buf) - 1, &tmp)) {
+ if (!*buf || *buf == '#' || *buf == '\n')
+ continue;
+ if (*buf == '[')
+ line_mode = parse_line_mode(buf, bus);
+ else if (patch_items[line_mode].parser)
+ patch_items[line_mode].parser(buf, bus, &codec);
+ }
+ release_firmware(fw);
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_load_patch);
+#endif /* CONFIG_SND_HDA_PATCH_LOADER */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 175f07a..20a66f8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -61,6 +61,9 @@ static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int probe_only[SNDRV_CARDS];
static int single_cmd;
static int enable_msi;
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+static char *patch[SNDRV_CARDS];
+#endif
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -84,6 +87,10 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
"(for debugging only).");
module_param(enable_msi, int, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+module_param_array(patch, charp, NULL, 0444);
+MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface.");
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
@@ -1331,8 +1338,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
[AZX_DRIVER_TERA] = 1,
};
-static int __devinit azx_codec_create(struct azx *chip, const char *model,
- int no_init)
+static int __devinit azx_codec_create(struct azx *chip, const char *model)
{
struct hda_bus_template bus_temp;
int c, codecs, err;
@@ -1391,7 +1397,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
for (c = 0; c < max_slots; c++) {
if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
struct hda_codec *codec;
- err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
+ err = snd_hda_codec_new(chip->bus, c, &codec);
if (err < 0)
continue;
codecs++;
@@ -1401,7 +1407,16 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
snd_printk(KERN_ERR SFX "no codecs initialized\n");
return -ENXIO;
}
+ return 0;
+}
+/* configure each codec instance */
+static int __devinit azx_codec_configure(struct azx *chip)
+{
+ struct hda_codec *codec;
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
+ snd_hda_codec_configure(codec);
+ }
return 0;
}
@@ -2284,6 +2299,30 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
}
}
+/*
+ * white-list for enable_msi
+ */
+static struct snd_pci_quirk msi_white_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
+ {}
+};
+
+static void __devinit check_msi(struct azx *chip)
+{
+ const struct snd_pci_quirk *q;
+
+ chip->msi = enable_msi;
+ if (chip->msi)
+ return;
+ q = snd_pci_quirk_lookup(chip->pci, msi_white_list);
+ if (q) {
+ printk(KERN_INFO
+ "hda_intel: msi for device %04x:%04x set to %d\n",
+ q->subvendor, q->subdevice, q->value);
+ chip->msi = q->value;
+ }
+}
+
/*
* constructor
@@ -2318,7 +2357,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->pci = pci;
chip->irq = -1;
chip->driver_type = driver_type;
- chip->msi = enable_msi;
+ check_msi(chip);
chip->dev_index = dev;
INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
@@ -2526,15 +2565,32 @@ static int __devinit azx_probe(struct pci_dev *pci,
return err;
}
+ /* set this here since it's referred in snd_hda_load_patch() */
+ snd_card_set_dev(card, &pci->dev);
+
err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
if (err < 0)
goto out_free;
card->private_data = chip;
/* create codec instances */
- err = azx_codec_create(chip, model[dev], probe_only[dev]);
+ err = azx_codec_create(chip, model[dev]);
if (err < 0)
goto out_free;
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+ if (patch[dev]) {
+ snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
+ patch[dev]);
+ err = snd_hda_load_patch(chip->bus, patch[dev]);
+ if (err < 0)
+ goto out_free;
+ }
+#endif
+ if (!probe_only[dev]) {
+ err = azx_codec_configure(chip);
+ if (err < 0)
+ goto out_free;
+ }
/* create PCM streams */
err = snd_hda_build_pcms(chip->bus);
@@ -2546,8 +2602,6 @@ static int __devinit azx_probe(struct pci_dev *pci,
if (err < 0)
goto out_free;
- snd_card_set_dev(card, &pci->dev);
-
err = snd_card_register(card);
if (err < 0)
goto out_free;
@@ -2649,11 +2703,15 @@ static struct pci_device_id azx_ids[] = {
/* this entry seems still valid -- i.e. without emu20kx chip */
{ PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC },
#endif
- /* AMD Generic, PCI class code and Vendor ID for HD Audio */
+ /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */
{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
.class_mask = 0xffffff,
.driver_data = AZX_DRIVER_GENERIC },
+ { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID),
+ .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+ .class_mask = 0xffffff,
+ .driver_data = AZX_DRIVER_GENERIC },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 8334901..5f1dcc5 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -99,7 +99,6 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char **slaves);
int snd_hda_codec_reset(struct hda_codec *codec);
-int snd_hda_codec_configure(struct hda_codec *codec);
/* amp value bits */
#define HDA_AMP_MUTE 0x80
@@ -408,6 +407,19 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
return codec->wcaps[nid - codec->start_nid];
}
+/* get the widget type from widget capability bits */
+#define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT)
+
+static inline unsigned int get_wcaps_channels(u32 wcaps)
+{
+ unsigned int chans;
+
+ chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+ chans = ((chans << 1) | 1) + 1;
+
+ return chans;
+}
+
u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 418c5d1..95f24e4 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -508,17 +508,14 @@ static void print_codec_info(struct snd_info_entry *entry,
unsigned int wid_caps =
snd_hda_param_read(codec, nid,
AC_PAR_AUDIO_WIDGET_CAP);
- unsigned int wid_type =
- (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ unsigned int wid_type = get_wcaps_type(wid_caps);
hda_nid_t conn[HDA_MAX_CONNECTIONS];
int conn_len = 0;
snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
get_wid_type_name(wid_type), wid_caps);
if (wid_caps & AC_WCAP_STEREO) {
- unsigned int chans;
- chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13;
- chans = ((chans << 1) | 1) + 1;
+ unsigned int chans = get_wcaps_channels(wid_caps);
if (chans == 2)
snd_iprintf(buffer, " Stereo");
else
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 403588c..215e72a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2982,7 +2982,8 @@ static int patch_ad1988(struct hda_codec *codec)
board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
ad1988_models, ad1988_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = AD1988_AUTO;
}
@@ -3702,19 +3703,29 @@ static struct hda_amp_list ad1884a_loopbacks[] = {
* Port F: Internal speakers
*/
-static struct hda_input_mux ad1884a_laptop_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 }, /* port-B */
- { "Internal Mic", 0x1 }, /* port-C */
- { "Dock Mic", 0x4 }, /* port-E */
- { "Mix", 0x3 },
- },
-};
+static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+ int mute = (!ucontrol->value.integer.value[0] &&
+ !ucontrol->value.integer.value[1]);
+ /* toggle GPIO1 according to the mute state */
+ snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ mute ? 0x02 : 0x0);
+ return ret;
+}
static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = ad1884a_mobile_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ },
HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
@@ -3729,36 +3740,9 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = ad198x_mux_enum_info,
- .get = ad198x_mux_enum_get,
- .put = ad198x_mux_enum_put,
- },
{ } /* end */
};
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- int mute = (!ucontrol->value.integer.value[0] &&
- !ucontrol->value.integer.value[1]);
- /* toggle GPIO1 according to the mute state */
- snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
- mute ? 0x02 : 0x0);
- return ret;
-}
-
static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
@@ -3828,6 +3812,63 @@ static int ad1884a_hp_init(struct hda_codec *codec)
return 0;
}
+/* mute internal speaker if HP or docking HP is plugged */
+static void ad1884a_laptop_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0);
+ present &= AC_PINSENSE_PRESENCE;
+ if (!present) {
+ present = snd_hda_codec_read(codec, 0x12, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present &= AC_PINSENSE_PRESENCE;
+ }
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ present ? 0x00 : 0x02);
+}
+
+/* switch to external mic if plugged */
+static void ad1884a_laptop_automic(struct hda_codec *codec)
+{
+ unsigned int idx;
+
+ if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE)
+ idx = 0;
+ else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE)
+ idx = 4;
+ else
+ idx = 1;
+ snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
+}
+
+/* unsolicited event for HP jack sensing */
+static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case AD1884A_HP_EVENT:
+ ad1884a_laptop_automute(codec);
+ break;
+ case AD1884A_MIC_EVENT:
+ ad1884a_laptop_automic(codec);
+ break;
+ }
+}
+
+/* initialize jack-sensing, too */
+static int ad1884a_laptop_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1884a_laptop_automute(codec);
+ ad1884a_laptop_automic(codec);
+ return 0;
+}
+
/* additional verbs for laptop model */
static struct hda_verb ad1884a_laptop_verbs[] = {
/* Port-A (HP) pin - always unmuted */
@@ -3844,11 +3885,19 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ /* Port-D (docking line-out) pin - default unmuted */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* analog mix */
{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* unsolicited event for pin-sense */
{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ /* allow to touch GPIO1 (for mute control) */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
{ } /* end */
};
@@ -4008,6 +4057,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
+ SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
{}
};
@@ -4057,9 +4107,8 @@ static int patch_ad1884a(struct hda_codec *codec)
spec->mixers[0] = ad1884a_laptop_mixers;
spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
spec->multiout.dig_out_nid = 0;
- spec->input_mux = &ad1884a_laptop_capture_source;
- codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
- codec->patch_ops.init = ad1884a_hp_init;
+ codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
+ codec->patch_ops.init = ad1884a_laptop_init;
/* set the upper-limit for mixer amp to 0dB for avoiding the
* possible damage by overloading
*/
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index 233e477..fb684f0 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec)
/* FIXME: we must check ELD and change the PCM parameters dynamically
*/
chans = get_wcaps(codec, CVT_NID);
- chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13;
- chans = ((chans << 1) | 1) + 1;
+ chans = get_wcaps_channels(chans);
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans;
return 0;
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 019ca7c..d08353d 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -459,8 +459,7 @@ static void parse_input(struct hda_codec *codec)
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
unsigned int wcaps = get_wcaps(codec, nid);
- unsigned int type = (wcaps & AC_WCAP_TYPE) >>
- AC_WCAP_TYPE_SHIFT;
+ unsigned int type = get_wcaps_type(wcaps);
if (type != AC_WID_AUD_IN)
continue;
if (snd_hda_get_connections(codec, nid, &pin, 1) != 1)
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
new file mode 100644
index 0000000..8ba3068
--- /dev/null
+++ b/sound/pci/hda/patch_cirrus.c
@@ -0,0 +1,1194 @@
+/*
+ * HD audio interface patch for Cirrus Logic CS420x chip
+ *
+ * Copyright (c) 2009 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/*
+ */
+
+struct cs_spec {
+ int board_config;
+ struct auto_pin_cfg autocfg;
+ struct hda_multi_out multiout;
+ struct snd_kcontrol *vmaster_sw;
+ struct snd_kcontrol *vmaster_vol;
+
+ hda_nid_t dac_nid[AUTO_CFG_MAX_OUTS];
+ hda_nid_t slave_dig_outs[2];
+
+ unsigned int input_idx[AUTO_PIN_LAST];
+ unsigned int capsrc_idx[AUTO_PIN_LAST];
+ hda_nid_t adc_nid[AUTO_PIN_LAST];
+ unsigned int adc_idx[AUTO_PIN_LAST];
+ unsigned int num_inputs;
+ unsigned int cur_input;
+ unsigned int automic_idx;
+ hda_nid_t cur_adc;
+ unsigned int cur_adc_stream_tag;
+ unsigned int cur_adc_format;
+ hda_nid_t dig_in;
+
+ struct hda_bind_ctls *capture_bind[2];
+
+ unsigned int gpio_mask;
+ unsigned int gpio_dir;
+ unsigned int gpio_data;
+
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+
+ unsigned int hp_detect:1;
+ unsigned int mic_detect:1;
+};
+
+/* available models */
+enum {
+ CS420X_MBP55,
+ CS420X_AUTO,
+ CS420X_MODELS
+};
+
+/* Vendor-specific processing widget */
+#define CS420X_VENDOR_NID 0x11
+#define CS_DIG_OUT1_PIN_NID 0x10
+#define CS_DIG_OUT2_PIN_NID 0x15
+#define CS_DMIC1_PIN_NID 0x12
+#define CS_DMIC2_PIN_NID 0x0e
+
+/* coef indices */
+#define IDX_SPDIF_STAT 0x0000
+#define IDX_SPDIF_CTL 0x0001
+#define IDX_ADC_CFG 0x0002
+/* SZC bitmask, 4 modes below:
+ * 0 = immediate,
+ * 1 = digital immediate, analog zero-cross
+ * 2 = digtail & analog soft-ramp
+ * 3 = digital soft-ramp, analog zero-cross
+ */
+#define CS_COEF_ADC_SZC_MASK (3 << 0)
+#define CS_COEF_ADC_MIC_SZC_MODE (3 << 0) /* SZC setup for mic */
+#define CS_COEF_ADC_LI_SZC_MODE (3 << 0) /* SZC setup for line-in */
+/* PGA mode: 0 = differential, 1 = signle-ended */
+#define CS_COEF_ADC_MIC_PGA_MODE (1 << 5) /* PGA setup for mic */
+#define CS_COEF_ADC_LI_PGA_MODE (1 << 6) /* PGA setup for line-in */
+#define IDX_DAC_CFG 0x0003
+/* SZC bitmask, 4 modes below:
+ * 0 = Immediate
+ * 1 = zero-cross
+ * 2 = soft-ramp
+ * 3 = soft-ramp on zero-cross
+ */
+#define CS_COEF_DAC_HP_SZC_MODE (3 << 0) /* nid 0x02 */
+#define CS_COEF_DAC_LO_SZC_MODE (3 << 2) /* nid 0x03 */
+#define CS_COEF_DAC_SPK_SZC_MODE (3 << 4) /* nid 0x04 */
+
+#define IDX_BEEP_CFG 0x0004
+/* 0x0008 - test reg key */
+/* 0x0009 - 0x0014 -> 12 test regs */
+/* 0x0015 - visibility reg */
+
+
+static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
+{
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_COEF_INDEX, idx);
+ return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+}
+
+static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
+ unsigned int coef)
+{
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_COEF_INDEX, idx);
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_PROC_COEF, coef);
+}
+
+
+#define HP_EVENT 1
+#define MIC_EVENT 2
+
+/*
+ * PCM callbacks
+ */
+static int cs_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int cs_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int cs_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int cs_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int cs_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int cs_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ spec->cur_adc = spec->adc_nid[spec->cur_input];
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+ snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+ return 0;
+}
+
+static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = 0;
+ return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream cs_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .open = cs_playback_pcm_open,
+ .prepare = cs_playback_pcm_prepare,
+ .cleanup = cs_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = cs_capture_pcm_prepare,
+ .cleanup = cs_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .open = cs_dig_playback_pcm_open,
+ .close = cs_dig_playback_pcm_close,
+ .prepare = cs_dig_playback_pcm_prepare,
+ .cleanup = cs_dig_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int cs_build_pcms(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->pcm_info = info;
+ codec->num_pcms = 0;
+
+ info->name = "Cirrus Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cs_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dac_nid[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->adc_nid[spec->cur_input];
+ codec->num_pcms++;
+
+ if (!spec->multiout.dig_out_nid && !spec->dig_in)
+ return 0;
+
+ info++;
+ info->name = "Cirrus Digital";
+ info->pcm_type = spec->autocfg.dig_out_type[0];
+ if (!info->pcm_type)
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ cs_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ cs_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+ }
+ codec->num_pcms++;
+
+ return 0;
+}
+
+/*
+ * parse codec topology
+ */
+
+static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ hda_nid_t dac;
+ if (!pin)
+ return 0;
+ if (snd_hda_get_connections(codec, pin, &dac, 1) != 1)
+ return 0;
+ return dac;
+}
+
+static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t pin = cfg->input_pins[idx];
+ unsigned int val = snd_hda_query_pin_caps(codec, pin);
+ if (!(val & AC_PINCAP_PRES_DETECT))
+ return 0;
+ val = snd_hda_codec_get_pincfg(codec, pin);
+ return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX);
+}
+
+static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int *idxp)
+{
+ int i;
+ hda_nid_t nid;
+
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ hda_nid_t pins[2];
+ unsigned int type;
+ int j, nums;
+ type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+ >> AC_WCAP_TYPE_SHIFT;
+ if (type != AC_WID_AUD_IN)
+ continue;
+ nums = snd_hda_get_connections(codec, nid, pins,
+ ARRAY_SIZE(pins));
+ if (nums <= 0)
+ continue;
+ for (j = 0; j < nums; j++) {
+ if (pins[j] == pin) {
+ *idxp = j;
+ return nid;
+ }
+ }
+ }
+ return 0;
+}
+
+static int is_active_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int val;
+ val = snd_hda_codec_get_pincfg(codec, nid);
+ return (get_defcfg_connect(val) != AC_JACK_PORT_NONE);
+}
+
+static int parse_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, extra_nids;
+ hda_nid_t dac;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ dac = get_dac(codec, cfg->line_out_pins[i]);
+ if (!dac)
+ break;
+ spec->dac_nid[i] = dac;
+ }
+ spec->multiout.num_dacs = i;
+ spec->multiout.dac_nids = spec->dac_nid;
+ spec->multiout.max_channels = i * 2;
+
+ /* add HP and speakers */
+ extra_nids = 0;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ dac = get_dac(codec, cfg->hp_pins[i]);
+ if (!dac)
+ break;
+ if (!i)
+ spec->multiout.hp_nid = dac;
+ else
+ spec->multiout.extra_out_nid[extra_nids++] = dac;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ dac = get_dac(codec, cfg->speaker_pins[i]);
+ if (!dac)
+ break;
+ spec->multiout.extra_out_nid[extra_nids++] = dac;
+ }
+
+ if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ cfg->speaker_outs = cfg->line_outs;
+ memcpy(cfg->speaker_pins, cfg->line_out_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->line_outs = 0;
+ }
+
+ return 0;
+}
+
+static int parse_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t pin = cfg->input_pins[i];
+ if (!pin)
+ continue;
+ spec->input_idx[spec->num_inputs] = i;
+ spec->capsrc_idx[i] = spec->num_inputs++;
+ spec->cur_input = i;
+ spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+ }
+ if (!spec->num_inputs)
+ return 0;
+
+ /* check whether the automatic mic switch is available */
+ if (spec->num_inputs == 2 &&
+ spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) {
+ if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) {
+ if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+ spec->mic_detect = 1;
+ spec->automic_idx = AUTO_PIN_FRONT_MIC;
+ }
+ } else {
+ if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+ spec->mic_detect = 1;
+ spec->automic_idx = AUTO_PIN_MIC;
+ }
+ }
+ }
+ return 0;
+}
+
+
+static int parse_digital_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid;
+
+ if (!cfg->dig_outs)
+ return 0;
+ if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1)
+ return 0;
+ spec->multiout.dig_out_nid = nid;
+ spec->multiout.share_spdif = 1;
+ if (cfg->dig_outs > 1 &&
+ snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) {
+ spec->slave_dig_outs[0] = nid;
+ codec->slave_dig_outs = spec->slave_dig_outs;
+ }
+ return 0;
+}
+
+static int parse_digital_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int idx;
+
+ if (cfg->dig_in_pin)
+ spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
+ return 0;
+}
+
+/*
+ * create mixer controls
+ */
+
+static const char *dir_sfx[2] = { "Playback", "Capture" };
+
+static int add_mute(struct hda_codec *codec, const char *name, int index,
+ unsigned int pval, int dir, struct snd_kcontrol **kctlp)
+{
+ char tmp[44];
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+ knew.private_value = pval;
+ snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]);
+ *kctlp = snd_ctl_new1(&knew, codec);
+ return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static int add_volume(struct hda_codec *codec, const char *name,
+ int index, unsigned int pval, int dir,
+ struct snd_kcontrol **kctlp)
+{
+ char tmp[32];
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+ knew.private_value = pval;
+ snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]);
+ *kctlp = snd_ctl_new1(&knew, codec);
+ return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac)
+{
+ unsigned int caps;
+
+ /* set the upper-limit for mixer amp to 0dB */
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ caps &= ~(0x7f << AC_AMPCAP_NUM_STEPS_SHIFT);
+ caps |= ((caps >> AC_AMPCAP_OFFSET_SHIFT) & 0x7f)
+ << AC_AMPCAP_NUM_STEPS_SHIFT;
+ snd_hda_override_amp_caps(codec, dac, HDA_OUTPUT, caps);
+}
+
+static int add_vmaster(struct hda_codec *codec, hda_nid_t dac)
+{
+ struct cs_spec *spec = codec->spec;
+ unsigned int tlv[4];
+ int err;
+
+ spec->vmaster_sw =
+ snd_ctl_make_virtual_master("Master Playback Switch", NULL);
+ err = snd_hda_ctl_add(codec, spec->vmaster_sw);
+ if (err < 0)
+ return err;
+
+ snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv);
+ spec->vmaster_vol =
+ snd_ctl_make_virtual_master("Master Playback Volume", tlv);
+ err = snd_hda_ctl_add(codec, spec->vmaster_vol);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
+ int num_ctls, int type)
+{
+ struct cs_spec *spec = codec->spec;
+ const char *name;
+ int err, index;
+ struct snd_kcontrol *kctl;
+ static char *speakers[] = {
+ "Front Speaker", "Surround Speaker", "Bass Speaker"
+ };
+ static char *line_outs[] = {
+ "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+ };
+
+ fix_volume_caps(codec, dac);
+ if (!spec->vmaster_sw) {
+ err = add_vmaster(codec, dac);
+ if (err < 0)
+ return err;
+ }
+
+ index = 0;
+ switch (type) {
+ case AUTO_PIN_HP_OUT:
+ name = "Headphone";
+ index = idx;
+ break;
+ case AUTO_PIN_SPEAKER_OUT:
+ if (num_ctls > 1)
+ name = speakers[idx];
+ else
+ name = "Speaker";
+ break;
+ default:
+ if (num_ctls > 1)
+ name = line_outs[idx];
+ else
+ name = "Line-Out";
+ break;
+ }
+
+ err = add_mute(codec, name, index,
+ HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+ if (err < 0)
+ return err;
+ err = snd_ctl_add_slave(spec->vmaster_sw, kctl);
+ if (err < 0)
+ return err;
+
+ err = add_volume(codec, name, index,
+ HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+ if (err < 0)
+ return err;
+ err = snd_ctl_add_slave(spec->vmaster_vol, kctl);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int build_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->line_out_pins[i]),
+ i, cfg->line_outs, cfg->line_out_type);
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < cfg->hp_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->hp_pins[i]),
+ i, cfg->hp_outs, AUTO_PIN_HP_OUT);
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->speaker_pins[i]),
+ i, cfg->speaker_outs, AUTO_PIN_SPEAKER_OUT);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ */
+
+static struct snd_kcontrol_new cs_capture_ctls[] = {
+ HDA_BIND_SW("Capture Switch", 0),
+ HDA_BIND_VOL("Capture Volume", 0),
+};
+
+static int change_cur_input(struct hda_codec *codec, unsigned int idx,
+ int force)
+{
+ struct cs_spec *spec = codec->spec;
+
+ if (spec->cur_input == idx && !force)
+ return 0;
+ if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) {
+ /* stream is running, let's swap the current ADC */
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = spec->adc_nid[idx];
+ snd_hda_codec_setup_stream(codec, spec->cur_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ }
+ snd_hda_codec_write(codec, spec->cur_adc, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ spec->adc_idx[idx]);
+ spec->cur_input = idx;
+ return 1;
+}
+
+static int cs_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ unsigned int idx;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = spec->num_inputs;
+ if (uinfo->value.enumerated.item >= spec->num_inputs)
+ uinfo->value.enumerated.item = spec->num_inputs - 1;
+ idx = spec->input_idx[uinfo->value.enumerated.item];
+ strcpy(uinfo->value.enumerated.name, auto_pin_cfg_labels[idx]);
+ return 0;
+}
+
+static int cs_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ ucontrol->value.enumerated.item[0] = spec->capsrc_idx[spec->cur_input];
+ return 0;
+}
+
+static int cs_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ unsigned int idx = ucontrol->value.enumerated.item[0];
+
+ if (idx >= spec->num_inputs)
+ return -EINVAL;
+ idx = spec->input_idx[idx];
+ return change_cur_input(codec, idx, 0);
+}
+
+static struct snd_kcontrol_new cs_capture_source = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = cs_capture_source_info,
+ .get = cs_capture_source_get,
+ .put = cs_capture_source_put,
+};
+
+static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
+ struct hda_ctl_ops *ops)
+{
+ struct cs_spec *spec = codec->spec;
+ struct hda_bind_ctls *bind;
+ int i, n;
+
+ bind = kzalloc(sizeof(*bind) + sizeof(long) * (spec->num_inputs + 1),
+ GFP_KERNEL);
+ if (!bind)
+ return NULL;
+ bind->ops = ops;
+ n = 0;
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!spec->adc_nid[i])
+ continue;
+ bind->values[n++] =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nid[i], 3,
+ spec->adc_idx[i], HDA_INPUT);
+ }
+ return bind;
+}
+
+static int build_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int i, err;
+
+ if (!spec->num_inputs)
+ return 0;
+
+ /* make bind-capture */
+ spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
+ spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
+ for (i = 0; i < 2; i++) {
+ struct snd_kcontrol *kctl;
+ if (!spec->capture_bind[i])
+ return -ENOMEM;
+ kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->private_value = (long)spec->capture_bind[i];
+ err = snd_hda_ctl_add(codec, kctl);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->num_inputs > 1 && !spec->mic_detect) {
+ err = snd_hda_ctl_add(codec,
+ snd_ctl_new1(&cs_capture_source, codec));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ */
+
+static int build_digital_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int err;
+
+ if (!spec->multiout.dig_out_nid)
+ return 0;
+
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int build_digital_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ if (spec->dig_in)
+ return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+ return 0;
+}
+
+/*
+ * auto-mute and auto-mic switching
+ */
+
+static void cs_automute(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int caps, present, hp_present;
+ hda_nid_t nid;
+ int i;
+
+ hp_present = 0;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ nid = cfg->hp_pins[i];
+ caps = snd_hda_query_pin_caps(codec, nid);
+ if (!(caps & AC_PINCAP_PRES_DETECT))
+ continue;
+ if (caps & AC_PINCAP_TRIG_REQ)
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ hp_present |= (present & AC_PINSENSE_PRESENCE) != 0;
+ if (hp_present)
+ break;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ nid = cfg->speaker_pins[i];
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ hp_present ? 0 : PIN_OUT);
+ }
+ if (spec->board_config == CS420X_MBP55) {
+ unsigned int gpio = hp_present ? 0x02 : 0x08;
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, gpio);
+ }
+}
+
+static void cs_automic(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid;
+ unsigned int caps, present;
+
+ nid = cfg->input_pins[spec->automic_idx];
+ caps = snd_hda_query_pin_caps(codec, nid);
+ if (caps & AC_PINCAP_TRIG_REQ)
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ if (present & AC_PINSENSE_PRESENCE)
+ change_cur_input(codec, spec->automic_idx, 0);
+ else {
+ unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ?
+ AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC;
+ change_cur_input(codec, imic, 0);
+ }
+}
+
+/*
+ */
+
+static void init_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ /* mute first */
+ for (i = 0; i < spec->multiout.num_dacs; i++)
+ snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ if (spec->multiout.hp_nid)
+ snd_hda_codec_write(codec, spec->multiout.hp_nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) {
+ if (!spec->multiout.extra_out_nid[i])
+ break;
+ snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ }
+
+ /* set appropriate pin controls */
+ for (i = 0; i < cfg->line_outs; i++)
+ snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ for (i = 0; i < cfg->hp_outs; i++) {
+ hda_nid_t nid = cfg->hp_pins[i];
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ if (!cfg->speaker_outs)
+ continue;
+ if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | HP_EVENT);
+ spec->hp_detect = 1;
+ }
+ }
+ for (i = 0; i < cfg->speaker_outs; i++)
+ snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ if (spec->hp_detect)
+ cs_automute(codec);
+}
+
+static void init_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int coef;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ unsigned int ctl;
+ hda_nid_t pin = cfg->input_pins[i];
+ if (!pin || !spec->adc_nid[i])
+ continue;
+ /* set appropriate pin control and mute first */
+ ctl = PIN_IN;
+ if (i <= AUTO_PIN_FRONT_MIC) {
+ unsigned int caps = snd_hda_query_pin_caps(codec, pin);
+ caps >>= AC_PINCAP_VREF_SHIFT;
+ if (caps & AC_PINCAP_VREF_80)
+ ctl = PIN_VREF80;
+ }
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ snd_hda_codec_write(codec, spec->adc_nid[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(spec->adc_idx[i]));
+ if (spec->mic_detect && spec->automic_idx == i)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | MIC_EVENT);
+ }
+ change_cur_input(codec, spec->cur_input, 1);
+ if (spec->mic_detect)
+ cs_automic(codec);
+
+ coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+ if (is_active_pin(codec, CS_DMIC2_PIN_NID))
+ coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
+ if (is_active_pin(codec, CS_DMIC1_PIN_NID))
+ coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0
+ * No effect if SPDIF_OUT2 is slected in
+ * IDX_SPDIF_CTL.
+ */
+ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+}
+
+static struct hda_verb cs_coef_init_verbs[] = {
+ {0x11, AC_VERB_SET_PROC_STATE, 1},
+ {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+ {0x11, AC_VERB_SET_PROC_COEF,
+ (0x002a /* DAC1/2/3 SZCMode Soft Ramp */
+ | 0x0040 /* Mute DACs on FIFO error */
+ | 0x1000 /* Enable DACs High Pass Filter */
+ | 0x0400 /* Disable Coefficient Auto increment */
+ )},
+ /* Beep */
+ {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */
+
+ {} /* terminator */
+};
+
+/* SPDIF setup */
+static void init_digital(struct hda_codec *codec)
+{
+ unsigned int coef;
+
+ coef = 0x0002; /* SRC_MUTE soft-mute on SPDIF (if no lock) */
+ coef |= 0x0008; /* Replace with mute on error */
+ if (is_active_pin(codec, CS_DIG_OUT2_PIN_NID))
+ coef |= 0x4000; /* RX to TX1 or TX2 Loopthru / SPDIF2
+ * SPDIF_OUT2 is shared with GPIO1 and
+ * DMIC_SDA2.
+ */
+ cs_vendor_coef_set(codec, IDX_SPDIF_CTL, coef);
+}
+
+static int cs_init(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+
+ snd_hda_sequence_write(codec, cs_coef_init_verbs);
+
+ if (spec->gpio_mask) {
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+ spec->gpio_mask);
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+ spec->gpio_dir);
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->gpio_data);
+ }
+
+ init_output(codec);
+ init_input(codec);
+ init_digital(codec);
+ return 0;
+}
+
+static int cs_build_controls(struct hda_codec *codec)
+{
+ int err;
+
+ err = build_output(codec);
+ if (err < 0)
+ return err;
+ err = build_input(codec);
+ if (err < 0)
+ return err;
+ err = build_digital_output(codec);
+ if (err < 0)
+ return err;
+ err = build_digital_input(codec);
+ if (err < 0)
+ return err;
+ return cs_init(codec);
+}
+
+static void cs_free(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ kfree(spec->capture_bind[0]);
+ kfree(spec->capture_bind[1]);
+ kfree(codec->spec);
+}
+
+static void cs_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch ((res >> 26) & 0x7f) {
+ case HP_EVENT:
+ cs_automute(codec);
+ break;
+ case MIC_EVENT:
+ cs_automic(codec);
+ break;
+ }
+}
+
+static struct hda_codec_ops cs_patch_ops = {
+ .build_controls = cs_build_controls,
+ .build_pcms = cs_build_pcms,
+ .init = cs_init,
+ .free = cs_free,
+ .unsol_event = cs_unsol_event,
+};
+
+static int cs_parse_auto_config(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+
+ err = parse_output(codec);
+ if (err < 0)
+ return err;
+ err = parse_input(codec);
+ if (err < 0)
+ return err;
+ err = parse_digital_output(codec);
+ if (err < 0)
+ return err;
+ err = parse_digital_input(codec);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static const char *cs420x_models[CS420X_MODELS] = {
+ [CS420X_MBP55] = "mbp55",
+ [CS420X_AUTO] = "auto",
+};
+
+
+static struct snd_pci_quirk cs420x_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+ {} /* terminator */
+};
+
+struct cs_pincfg {
+ hda_nid_t nid;
+ u32 val;
+};
+
+static struct cs_pincfg mbp55_pincfgs[] = {
+ { 0x09, 0x012b4030 },
+ { 0x0a, 0x90100121 },
+ { 0x0b, 0x90100120 },
+ { 0x0c, 0x400000f0 },
+ { 0x0d, 0x90a00110 },
+ { 0x0e, 0x400000f0 },
+ { 0x0f, 0x400000f0 },
+ { 0x10, 0x014be040 },
+ { 0x12, 0x400000f0 },
+ { 0x15, 0x400000f0 },
+ {} /* terminator */
+};
+
+static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
+ [CS420X_MBP55] = mbp55_pincfgs,
+};
+
+static void fix_pincfg(struct hda_codec *codec, int model)
+{
+ const struct cs_pincfg *cfg = cs_pincfgs[model];
+ if (!cfg)
+ return;
+ for (; cfg->nid; cfg++)
+ snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+
+
+static int patch_cs420x(struct hda_codec *codec)
+{
+ struct cs_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ spec->board_config =
+ snd_hda_check_board_config(codec, CS420X_MODELS,
+ cs420x_models, cs420x_cfg_tbl);
+ if (spec->board_config >= 0)
+ fix_pincfg(codec, spec->board_config);
+
+ switch (spec->board_config) {
+ case CS420X_MBP55:
+ /* GPIO1 = headphones */
+ /* GPIO3 = speakers */
+ spec->gpio_mask = 0x0a;
+ spec->gpio_dir = 0x0a;
+ break;
+ }
+
+ err = cs_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
+
+ codec->patch_ops = cs_patch_ops;
+
+ return 0;
+
+ error:
+ kfree(codec->spec);
+ codec->spec = NULL;
+ return err;
+}
+
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_cirrus[] = {
+ { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
+ { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
+ {} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:10134206");
+MODULE_ALIAS("snd-hda-codec-id:10134207");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
+
+static struct hda_codec_preset_list cirrus_list = {
+ .preset = snd_hda_preset_cirrus,
+ .owner = THIS_MODULE,
+};
+
+static int __init patch_cirrus_init(void)
+{
+ return snd_hda_add_codec_preset(&cirrus_list);
+}
+
+static void __exit patch_cirrus_exit(void)
+{
+ snd_hda_delete_codec_preset(&cirrus_list);
+}
+
+module_init(patch_cirrus_init)
+module_exit(patch_cirrus_exit)
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c921264..780e1a7 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -635,7 +635,8 @@ static int patch_cmi9880(struct hda_codec *codec)
cmi9880_models,
cmi9880_cfg_tbl);
if (spec->board_config < 0) {
- snd_printdd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
spec->board_config = CMI_AUTO; /* try everything */
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ac868c5..9d899ed 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -108,6 +108,8 @@ struct conexant_spec {
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+ unsigned int dell_automute;
+ unsigned int port_d_mode;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec)
return 0;
}
+/* Conexant 5066 specific */
+
+static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
+static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
+static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
+#define CXT5066_SPDIF_OUT 0x21
+
+static struct hda_channel_mode cxt5066_modes[1] = {
+ { 2, NULL },
+};
+
+static void cxt5066_update_speaker(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int pinctl;
+
+ snd_printdd("CXT5066: update speaker, hp_present=%d\n",
+ spec->hp_present);
+
+ /* Port A (HP) */
+ pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0;
+ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl);
+
+ /* Port D (HP/LO) */
+ pinctl = ((spec->hp_present & 2) && spec->cur_eapd)
+ ? spec->port_d_mode : 0;
+ snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl);
+
+ /* CLASS_D AMP */
+ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
+ snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl);
+
+ if (spec->dell_automute) {
+ /* DELL AIO Port Rule: PortA > PortD > IntSpk */
+ pinctl = (!(spec->hp_present & 1) && spec->cur_eapd)
+ ? PIN_OUT : 0;
+ snd_hda_codec_write(codec, 0x1c, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+ }
+}
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (!cxt_eapd_put(kcontrol, ucontrol))
+ return 0;
+
+ cxt5066_update_speaker(codec);
+ return 1;
+}
+
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5066_automic(struct hda_codec *codec)
+{
+ static struct hda_verb ext_mic_present[] = {
+ /* enable external mic, port B */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* switch to external mic input */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* disable internal mic, port C */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {}
+ };
+ static struct hda_verb ext_mic_absent[] = {
+ /* enable internal mic, port C */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* switch to internal mic input */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 1},
+
+ /* disable external mic, port B */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {}
+ };
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x1a, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ if (present) {
+ snd_printdd("CXT5066: external microphone detected\n");
+ snd_hda_sequence_write(codec, ext_mic_present);
+ } else {
+ snd_printdd("CXT5066: external microphone absent\n");
+ snd_hda_sequence_write(codec, ext_mic_absent);
+ }
+}
+
+/* mute internal speaker if HP is plugged */
+static void cxt5066_hp_automute(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int portA, portD;
+
+ /* Port A */
+ portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ /* Port D */
+ portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE) << 1;
+
+ spec->hp_present = !!(portA | portD);
+ snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n",
+ portA, portD, spec->hp_present);
+ cxt5066_update_speaker(codec);
+}
+
+/* unsolicited event for jack sensing */
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5066_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5066_automic(codec);
+ break;
+ }
+}
+
+static const struct hda_input_mux cxt5066_analog_mic_boost = {
+ .num_items = 5,
+ .items = {
+ { "0dB", 0 },
+ { "10dB", 1 },
+ { "20dB", 2 },
+ { "30dB", 3 },
+ { "40dB", 4 },
+ },
+};
+
+static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo);
+}
+
+static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int val;
+
+ val = snd_hda_codec_read(codec, 0x17, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT);
+
+ ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
+ return 0;
+}
+
+static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
+ unsigned int idx;
+
+ if (!imux->num_items)
+ return 0;
+ idx = ucontrol->value.enumerated.item[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+
+ snd_hda_codec_write_cache(codec, 0x17, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+ imux->items[idx].index);
+
+ return 1;
+}
+
+static struct hda_input_mux cxt5066_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic B", 0 },
+ { "Mic C", 1 },
+ { "Mic E", 2 },
+ { "Mic F", 3 },
+ },
+};
+
+static struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new cxt5066_mixer_master[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+ {}
+};
+
+static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
+ .info = snd_hda_mixer_amp_volume_info,
+ .get = snd_hda_mixer_amp_volume_get,
+ .put = snd_hda_mixer_amp_volume_put,
+ .tlv = { .c = snd_hda_mixer_amp_tlv },
+ /* offset by 28 volume steps to limit minimum gain to -46dB */
+ .private_value =
+ HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28),
+ },
+ {}
+};
+
+static struct snd_kcontrol_new cxt5066_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5066_hp_master_sw_put,
+ .private_value = 0x1d,
+ },
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Mic Boost Capture Enum",
+ .info = cxt5066_mic_boost_mux_enum_info,
+ .get = cxt5066_mic_boost_mux_enum_get,
+ .put = cxt5066_mic_boost_mux_enum_put,
+ },
+
+ HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
+ HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others),
+ {}
+};
+
+static struct hda_verb cxt5066_init_verbs[] = {
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
+ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
+
+ /* Speakers */
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* HP, Amp */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* DAC1 */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* no digital microphone support yet */
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Audio input selector */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
+
+ /* SPDIF route: PCM */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* EAPD */
+ {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+
+ /* not handling these yet */
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+ { } /* end */
+};
+
+static struct hda_verb cxt5066_init_verbs_olpc[] = {
+ /* Port A: headphones */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* Port B: external microphone */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* Port C: internal microphone */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* Port D: unused */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port E: unused, but has primary EAPD */
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+
+ /* Port F: unused */
+ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port G: internal speakers */
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* DAC1 */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* DAC2: unused */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+ /* Disable digital microphone port */
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Audio input selectors */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+
+ /* Disable SPDIF */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* enable unsolicited events for Port A and B */
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
+static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { } /* end */
+};
+
+/* initialize jack-sensing, too */
+static int cxt5066_init(struct hda_codec *codec)
+{
+ snd_printdd("CXT5066: init\n");
+ conexant_init(codec);
+ if (codec->patch_ops.unsol_event) {
+ cxt5066_hp_automute(codec);
+ cxt5066_automic(codec);
+ }
+ return 0;
+}
+
+enum {
+ CXT5066_LAPTOP, /* Laptops w/ EAPD support */
+ CXT5066_DELL_LAPTOP, /* Dell Laptop */
+ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */
+ CXT5066_MODELS
+};
+
+static const char *cxt5066_models[CXT5066_MODELS] = {
+ [CXT5066_LAPTOP] = "laptop",
+ [CXT5066_DELL_LAPTOP] = "dell-laptop",
+ [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5",
+};
+
+static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
+ CXT5066_LAPTOP),
+ SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
+ CXT5066_DELL_LAPTOP),
+ {}
+};
+
+static int patch_cxt5066(struct hda_codec *codec)
+{
+ struct conexant_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ codec->patch_ops = conexant_patch_ops;
+ codec->patch_ops.init = cxt5066_init;
+
+ spec->dell_automute = 0;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
+ spec->multiout.dac_nids = cxt5066_dac_nids;
+ spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = cxt5066_adc_nids;
+ spec->capsrc_nids = cxt5066_capsrc_nids;
+ spec->input_mux = &cxt5066_capture_source;
+
+ spec->port_d_mode = PIN_HP;
+
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = cxt5066_init_verbs;
+ spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes);
+ spec->channel_mode = cxt5066_modes;
+ spec->cur_adc = 0;
+ spec->cur_adc_idx = 0;
+
+ board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
+ cxt5066_models, cxt5066_cfg_tbl);
+ switch (board_config) {
+ default:
+ case CXT5066_LAPTOP:
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ break;
+ case CXT5066_DELL_LAPTOP:
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+
+ spec->port_d_mode = PIN_OUT;
+ spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo;
+ spec->num_init_verbs++;
+ spec->dell_automute = 1;
+ break;
+ case CXT5066_OLPC_XO_1_5:
+ codec->patch_ops.unsol_event = cxt5066_unsol_event;
+ spec->init_verbs[0] = cxt5066_init_verbs_olpc;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ spec->port_d_mode = 0;
+
+ /* no S/PDIF out */
+ spec->multiout.dig_out_nid = 0;
+
+ /* input source automatically selected */
+ spec->input_mux = NULL;
+ break;
+ }
+
+ return 0;
+}
/*
*/
@@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5047 },
{ .id = 0x14f15051, .name = "CX20561 (Hermosa)",
.patch = patch_cxt5051 },
+ { .id = 0x14f15066, .name = "CX20582 (Pebble)",
+ .patch = patch_cxt5066 },
{} /* terminator */
};
MODULE_ALIAS("snd-hda-codec-id:14f15045");
MODULE_ALIAS("snd-hda-codec-id:14f15047");
MODULE_ALIAS("snd-hda-codec-id:14f15051");
+MODULE_ALIAS("snd-hda-codec-id:14f15066");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index fcc77fe..01a18ed 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -33,8 +33,8 @@
#include "hda_codec.h"
#include "hda_local.h"
-#define CVT_NID 0x02 /* audio converter */
-#define PIN_NID 0x03 /* HDMI output pin */
+static hda_nid_t cvt_nid; /* audio converter */
+static hda_nid_t pin_nid; /* HDMI output pin */
#define INTEL_HDMI_EVENT_TAG 0x08
@@ -44,30 +44,6 @@ struct intel_hdmi_spec {
struct hdmi_eld sink_eld;
};
-static struct hda_verb pinout_enable_verb[] = {
- {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {} /* terminator */
-};
-
-static struct hda_verb unsolicited_response_verb[] = {
- {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
- INTEL_HDMI_EVENT_TAG},
- {}
-};
-
-static struct hda_verb def_chan_map[] = {
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66},
- {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77},
- {}
-};
-
-
struct hdmi_audio_infoframe {
u8 type; /* 0x84 */
u8 ver; /* 0x01 */
@@ -244,11 +220,12 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
static void hdmi_enable_output(struct hda_codec *codec)
{
/* Unmute */
- if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
- snd_hda_codec_write(codec, PIN_NID, 0,
+ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, pin_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
/* Enable pin out */
- snd_hda_sequence_write(codec, pinout_enable_verb);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
}
/*
@@ -256,8 +233,8 @@ static void hdmi_enable_output(struct hda_codec *codec)
*/
static void hdmi_start_infoframe_trans(struct hda_codec *codec)
{
- hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
- snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+ hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
+ snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
AC_DIPXMIT_BEST);
}
@@ -266,20 +243,20 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec)
*/
static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
{
- hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
- snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+ hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
+ snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
AC_DIPXMIT_DISABLE);
}
static int hdmi_get_channel_count(struct hda_codec *codec)
{
- return 1 + snd_hda_codec_read(codec, CVT_NID, 0,
+ return 1 + snd_hda_codec_read(codec, cvt_nid, 0,
AC_VERB_GET_CVT_CHAN_COUNT, 0);
}
static void hdmi_set_channel_count(struct hda_codec *codec, int chs)
{
- snd_hda_codec_write(codec, CVT_NID, 0,
+ snd_hda_codec_write(codec, cvt_nid, 0,
AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
if (chs != hdmi_get_channel_count(codec))
@@ -294,7 +271,7 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec)
int slot;
for (i = 0; i < 8; i++) {
- slot = snd_hda_codec_read(codec, CVT_NID, 0,
+ slot = snd_hda_codec_read(codec, cvt_nid, 0,
AC_VERB_GET_HDMI_CHAN_SLOT, i);
printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n",
slot >> 4, slot & 0x7);
@@ -307,7 +284,7 @@ static void hdmi_parse_eld(struct hda_codec *codec)
struct intel_hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->sink_eld;
- if (!snd_hdmi_get_eld(eld, codec, PIN_NID))
+ if (!snd_hdmi_get_eld(eld, codec, pin_nid))
snd_hdmi_show_eld(eld);
}
@@ -322,11 +299,11 @@ static void hdmi_debug_dip_size(struct hda_codec *codec)
int i;
int size;
- size = snd_hdmi_get_eld_size(codec, PIN_NID);
+ size = snd_hdmi_get_eld_size(codec, pin_nid);
printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size);
for (i = 0; i < 8; i++) {
- size = snd_hda_codec_read(codec, PIN_NID, 0,
+ size = snd_hda_codec_read(codec, pin_nid, 0,
AC_VERB_GET_HDMI_DIP_SIZE, i);
printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size);
}
@@ -340,15 +317,15 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec)
int size;
int pi, bi;
for (i = 0; i < 8; i++) {
- size = snd_hda_codec_read(codec, PIN_NID, 0,
+ size = snd_hda_codec_read(codec, pin_nid, 0,
AC_VERB_GET_HDMI_DIP_SIZE, i);
if (size == 0)
continue;
- hdmi_set_dip_index(codec, PIN_NID, i, 0x0);
+ hdmi_set_dip_index(codec, pin_nid, i, 0x0);
for (j = 1; j < 1000; j++) {
- hdmi_write_dip_byte(codec, PIN_NID, 0x0);
- hdmi_get_dip_index(codec, PIN_NID, &pi, &bi);
+ hdmi_write_dip_byte(codec, pin_nid, 0x0);
+ hdmi_get_dip_index(codec, pin_nid, &pi, &bi);
if (pi != i)
snd_printd(KERN_INFO "dip index %d: %d != %d\n",
bi, pi, i);
@@ -376,9 +353,9 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
sum += params[i];
ai->checksum = - sum;
- hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+ hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
for (i = 0; i < sizeof(ai); i++)
- hdmi_write_dip_byte(codec, PIN_NID, params[i]);
+ hdmi_write_dip_byte(codec, pin_nid, params[i]);
}
/*
@@ -465,6 +442,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
static void hdmi_setup_channel_mapping(struct hda_codec *codec,
struct hdmi_audio_infoframe *ai)
{
+ int i;
+
if (!ai->CA)
return;
@@ -473,7 +452,11 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec,
* ALSA sequence is front/surr/clfe/side?
*/
- snd_hda_sequence_write(codec, def_chan_map);
+ for (i = 0; i < 8; i++)
+ snd_hda_codec_write(codec, cvt_nid, 0,
+ AC_VERB_SET_HDMI_CHAN_SLOT,
+ (i << 4) | i);
+
hdmi_debug_channel_mapping(codec);
}
@@ -597,7 +580,6 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
- .nid = CVT_NID, /* NID to query formats and rates and setup streams */
.ops = {
.open = intel_hdmi_playback_pcm_open,
.close = intel_hdmi_playback_pcm_close,
@@ -613,6 +595,9 @@ static int intel_hdmi_build_pcms(struct hda_codec *codec)
codec->num_pcms = 1;
codec->pcm_info = info;
+ /* NID to query formats and rates and setup streams */
+ intel_hdmi_pcm_playback.nid = cvt_nid;
+
info->name = "INTEL HDMI";
info->pcm_type = HDA_PCM_TYPE_HDMI;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback;
@@ -636,8 +621,9 @@ static int intel_hdmi_init(struct hda_codec *codec)
{
hdmi_enable_output(codec);
- snd_hda_sequence_write(codec, unsolicited_response_verb);
-
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | INTEL_HDMI_EVENT_TAG);
return 0;
}
@@ -657,7 +643,7 @@ static struct hda_codec_ops intel_hdmi_patch_ops = {
.unsol_event = intel_hdmi_unsol_event,
};
-static int patch_intel_hdmi(struct hda_codec *codec)
+static int do_patch_intel_hdmi(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec;
@@ -667,7 +653,7 @@ static int patch_intel_hdmi(struct hda_codec *codec)
spec->multiout.num_dacs = 0; /* no analog */
spec->multiout.max_channels = 8;
- spec->multiout.dig_out_nid = CVT_NID;
+ spec->multiout.dig_out_nid = cvt_nid;
codec->spec = spec;
codec->patch_ops = intel_hdmi_patch_ops;
@@ -679,12 +665,27 @@ static int patch_intel_hdmi(struct hda_codec *codec)
return 0;
}
+static int patch_intel_hdmi(struct hda_codec *codec)
+{
+ cvt_nid = 0x02;
+ pin_nid = 0x03;
+ return do_patch_intel_hdmi(codec);
+}
+
+static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec)
+{
+ cvt_nid = 0x02;
+ pin_nid = 0x04;
+ return do_patch_intel_hdmi(codec);
+}
+
static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
{ .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi },
{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
+ { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak },
{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi },
{} /* terminator */
};
@@ -694,6 +695,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862801");
MODULE_ALIAS("snd-hda-codec-id:80862802");
MODULE_ALIAS("snd-hda-codec-id:80862803");
MODULE_ALIAS("snd-hda-codec-id:80862804");
+MODULE_ALIAS("snd-hda-codec-id:80860054");
MODULE_ALIAS("snd-hda-codec-id:10951392");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index f5792e2..c8435c9 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -377,6 +377,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
*/
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -385,6 +386,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
};
MODULE_ALIAS("snd-hda-codec-id:10de0002");
+MODULE_ALIAS("snd-hda-codec-id:10de0003");
MODULE_ALIAS("snd-hda-codec-id:10de0006");
MODULE_ALIAS("snd-hda-codec-id:10de0007");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 30eeb30..7ed47f6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -208,12 +208,6 @@ enum {
ALC885_MBP3,
ALC885_MB5,
ALC885_IMAC24,
- ALC882_AUTO,
- ALC882_MODEL_LAST,
-};
-
-/* ALC883 models */
-enum {
ALC883_3ST_2ch_DIG,
ALC883_3ST_6ch_DIG,
ALC883_3ST_6ch,
@@ -226,6 +220,7 @@ enum {
ALC888_ACER_ASPIRE_4930G,
ALC888_ACER_ASPIRE_6530G,
ALC888_ACER_ASPIRE_8930G,
+ ALC888_ACER_ASPIRE_7730G,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
@@ -237,17 +232,20 @@ enum {
ALC888_3ST_HP,
ALC888_6ST_DELL,
ALC883_MITAC,
+ ALC883_CLEVO_M540R,
ALC883_CLEVO_M720,
ALC883_FUJITSU_PI2515,
ALC888_FUJITSU_XA3530,
ALC883_3ST_6ch_INTEL,
+ ALC889A_INTEL,
+ ALC889_INTEL,
ALC888_ASUS_M90V,
ALC888_ASUS_EEE1601,
ALC889A_MB31,
ALC1200_ASUS_P5Q,
ALC883_SONY_VAIO_TT,
- ALC883_AUTO,
- ALC883_MODEL_LAST,
+ ALC882_AUTO,
+ ALC882_MODEL_LAST,
};
/* for GPIO Poll */
@@ -262,6 +260,14 @@ enum {
ALC_INIT_GPIO3,
};
+struct alc_mic_route {
+ hda_nid_t pin;
+ unsigned char mux_idx;
+ unsigned char amix_idx;
+};
+
+#define MUX_IDX_UNDEF ((unsigned char)-1)
+
struct alc_spec {
/* codec parameterization */
struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
@@ -304,6 +310,8 @@ struct alc_spec {
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
unsigned int cur_mux[3];
+ struct alc_mic_route ext_mic;
+ struct alc_mic_route int_mic;
/* channel model */
const struct hda_channel_mode *channel_mode;
@@ -320,6 +328,8 @@ struct alc_spec {
struct snd_array kctls;
struct hda_input_mux private_imux[3];
hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+ hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS];
+ hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS];
/* hooks */
void (*init_hook)(struct hda_codec *codec);
@@ -329,6 +339,7 @@ struct alc_spec {
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
unsigned int master_sw: 1;
+ unsigned int auto_mic:1;
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
@@ -370,6 +381,7 @@ struct alc_config_preset {
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
+ void (*setup)(struct hda_codec *);
void (*init_hook)(struct hda_codec *);
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_amp_list *loopbacks;
@@ -417,7 +429,7 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
- type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_AUD_MIX) {
/* Matrix-mixer style (e.g. ALC882) */
unsigned int *cur_val = &spec->cur_mux[adc_idx];
@@ -842,9 +854,10 @@ static void print_realtek_coef(struct snd_info_buffer *buffer,
/*
* set up from the preset table
*/
-static void setup_preset(struct alc_spec *spec,
+static void setup_preset(struct hda_codec *codec,
const struct alc_config_preset *preset)
{
+ struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
@@ -886,6 +899,9 @@ static void setup_preset(struct alc_spec *spec,
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = preset->loopbacks;
#endif
+
+ if (preset->setup)
+ preset->setup(codec);
}
/* Enable GPIO mask and set output */
@@ -965,30 +981,64 @@ static void alc_automute_pin(struct hda_codec *codec)
}
}
-#if 0 /* it's broken in some cases -- temporarily disabled */
+static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
+ hda_nid_t nid)
+{
+ hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+ int i, nums;
+
+ nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+ for (i = 0; i < nums; i++)
+ if (conn[i] == nid)
+ return i;
+ return -1;
+}
+
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int present;
- unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
- unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
- unsigned int mix_nid = spec->capsrc_nids[0];
- unsigned int capsrc_idx_mic, capsrc_idx_fmic;
-
- capsrc_idx_mic = mic_nid - 0x18;
- capsrc_idx_fmic = fmic_nid - 0x18;
- present = snd_hda_codec_read(codec, mic_nid, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0));
- snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ struct alc_mic_route *dead, *alive;
+ unsigned int present, type;
+ hda_nid_t cap_nid;
+
+ if (!spec->auto_mic)
+ return;
+ if (!spec->int_mic.pin || !spec->ext_mic.pin)
+ return;
+ if (snd_BUG_ON(!spec->adc_nids))
+ return;
+
+ cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
+
+ present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present &= AC_PINSENSE_PRESENCE;
+ if (present) {
+ alive = &spec->ext_mic;
+ dead = &spec->int_mic;
+ } else {
+ alive = &spec->int_mic;
+ dead = &spec->ext_mic;
+ }
+
+ type = get_wcaps_type(get_wcaps(codec, cap_nid));
+ if (type == AC_WID_AUD_MIX) {
+ /* Matrix-mixer style (e.g. ALC882) */
+ snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+ alive->mux_idx,
+ HDA_AMP_MUTE, 0);
+ snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+ dead->mux_idx,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* MUX style (e.g. ALC880) */
+ snd_hda_codec_write_cache(codec, cap_nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ alive->mux_idx);
+ }
+
+ /* FIXME: analog mixer */
}
-#else
-#define alc_mic_automute(codec) do {} while(0) /* NOP */
-#endif /* disabled */
/* unsolicited event for HP jack sensing */
static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1031,6 +1081,16 @@ static void alc888_coef_init(struct hda_codec *codec)
AC_VERB_SET_PROC_COEF, 0x3030);
}
+static void alc889_coef_init(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
+}
+
static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
unsigned int tmp;
@@ -1088,15 +1148,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
case 0x10ec0885:
case 0x10ec0887:
case 0x10ec0889:
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_COEF_INDEX, 7);
- tmp = snd_hda_codec_read(codec, 0x20, 0,
- AC_VERB_GET_PROC_COEF, 0);
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_COEF_INDEX, 7);
- snd_hda_codec_write(codec, 0x20, 0,
- AC_VERB_SET_PROC_COEF,
- tmp | 0x2010);
+ alc889_coef_init(codec);
break;
case 0x10ec0888:
alc888_coef_init(codec);
@@ -1142,6 +1194,55 @@ static void alc_init_auto_hp(struct hda_codec *codec)
spec->unsol_event = alc_sku_unsol_event;
}
+static void alc_init_auto_mic(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t fixed, ext;
+ int i;
+
+ /* there must be only two mic inputs exclusively */
+ for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++)
+ if (cfg->input_pins[i])
+ return;
+
+ fixed = ext = 0;
+ for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) {
+ hda_nid_t nid = cfg->input_pins[i];
+ unsigned int defcfg;
+ if (!nid)
+ return;
+ defcfg = snd_hda_codec_get_pincfg(codec, nid);
+ switch (get_defcfg_connect(defcfg)) {
+ case AC_JACK_PORT_FIXED:
+ if (fixed)
+ return; /* already occupied */
+ fixed = nid;
+ break;
+ case AC_JACK_PORT_COMPLEX:
+ if (ext)
+ return; /* already occupied */
+ ext = nid;
+ break;
+ default:
+ return; /* invalid entry */
+ }
+ }
+ if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+ return; /* no unsol support */
+ snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
+ ext, fixed);
+ spec->ext_mic.pin = ext;
+ spec->int_mic.pin = fixed;
+ spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
+ spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
+ spec->auto_mic = 1;
+ snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC880_MIC_EVENT);
+ spec->unsol_event = alc_sku_unsol_event;
+}
+
/* check subsystem ID and set up device-specific initialization;
* return 1 if initialized, 0 if invalid SSID
*/
@@ -1243,6 +1344,7 @@ do_sku:
}
alc_init_auto_hp(codec);
+ alc_init_auto_mic(codec);
return 1;
}
@@ -1255,6 +1357,7 @@ static void alc_ssid_check(struct hda_codec *codec,
"Enable default setup for auto mode as fallback\n");
spec->init_amp = ALC_INIT_DEFAULT;
alc_init_auto_hp(codec);
+ alc_init_auto_mic(codec);
}
}
@@ -1436,7 +1539,25 @@ static void alc_automute_amp_unsol_event(struct hda_codec *codec,
alc_automute_amp(codec);
}
-static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
+static void alc889_automute_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
+ spec->autocfg.speaker_pins[3] = 0x19;
+ spec->autocfg.speaker_pins[4] = 0x1a;
+}
+
+static void alc889_intel_init_hook(struct hda_codec *codec)
+{
+ alc889_coef_init(codec);
+ alc_automute_amp(codec);
+}
+
+static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1444,7 +1565,6 @@ static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
spec->autocfg.hp_pins[1] = 0x1b; /* hp */
spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
spec->autocfg.speaker_pins[1] = 0x15; /* bass */
- alc_automute_amp(codec);
}
/*
@@ -1643,16 +1763,15 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
{ } /* end */
};
-static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec)
+static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_amp(codec);
}
-static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec)
+static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1660,10 +1779,9 @@ static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x17;
- alc_automute_amp(codec);
}
-static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec)
+static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1671,7 +1789,6 @@ static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x1b;
- alc_automute_amp(codec);
}
/*
@@ -2651,13 +2768,17 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec)
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
-static void alc880_uniwill_init_hook(struct hda_codec *codec)
+static void alc880_uniwill_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x16;
+}
+
+static void alc880_uniwill_init_hook(struct hda_codec *codec)
+{
alc_automute_amp(codec);
alc880_uniwill_mic_automute(codec);
}
@@ -2678,13 +2799,12 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
}
}
-static void alc880_uniwill_p53_init_hook(struct hda_codec *codec)
+static void alc880_uniwill_p53_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_automute_amp(codec);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -2947,13 +3067,12 @@ static struct hda_verb alc880_lg_init_verbs[] = {
};
/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_init_hook(struct hda_codec *codec)
+static void alc880_lg_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x17;
- alc_automute_amp(codec);
}
/*
@@ -3032,13 +3151,12 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = {
};
/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_lw_init_hook(struct hda_codec *codec)
+static void alc880_lg_lw_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_amp(codec);
}
static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -3104,13 +3222,12 @@ static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
alc880_medion_rim_automute(codec);
}
-static void alc880_medion_rim_init_hook(struct hda_codec *codec)
+static void alc880_medion_rim_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- alc880_medion_rim_automute(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -3977,7 +4094,8 @@ static struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_f1734_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_init_hook,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_automute_amp,
},
[ALC880_ASUS] = {
.mixers = { alc880_asus_mixer },
@@ -4054,6 +4172,7 @@ static struct alc_config_preset alc880_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_unsol_event,
+ .setup = alc880_uniwill_setup,
.init_hook = alc880_uniwill_init_hook,
},
[ALC880_UNIWILL_P53] = {
@@ -4066,7 +4185,8 @@ static struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_init_hook,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_automute_amp,
},
[ALC880_FUJITSU] = {
.mixers = { alc880_fujitsu_mixer },
@@ -4080,7 +4200,8 @@ static struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_init_hook,
+ .setup = alc880_uniwill_p53_setup,
+ .init_hook = alc_automute_amp,
},
[ALC880_CLEVO] = {
.mixers = { alc880_three_stack_mixer },
@@ -4106,7 +4227,8 @@ static struct alc_config_preset alc880_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc880_lg_init_hook,
+ .setup = alc880_lg_setup,
+ .init_hook = alc_automute_amp,
#ifdef CONFIG_SND_HDA_POWER_SAVE
.loopbacks = alc880_lg_loopbacks,
#endif
@@ -4122,7 +4244,8 @@ static struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_lg_lw_modes,
.input_mux = &alc880_lg_lw_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc880_lg_lw_init_hook,
+ .setup = alc880_lg_lw_setup,
+ .init_hook = alc_automute_amp,
},
[ALC880_MEDION_RIM] = {
.mixers = { alc880_medion_rim_mixer },
@@ -4136,7 +4259,8 @@ static struct alc_config_preset alc880_presets[] = {
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_medion_rim_capture_source,
.unsol_event = alc880_medion_rim_unsol_event,
- .init_hook = alc880_medion_rim_init_hook,
+ .setup = alc880_medion_rim_setup,
+ .init_hook = alc880_medion_rim_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = {
@@ -4189,8 +4313,6 @@ static int add_control(struct alc_spec *spec, int type, const char *name,
#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
-#define alc880_is_input_pin(nid) ((nid) >= 0x18)
-#define alc880_input_pin_idx(nid) ((nid) - 0x18)
#define alc880_idx_to_dac(nid) ((nid) + 0x02)
#define alc880_dac_to_idx(nid) ((nid) - 0x02)
#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
@@ -4278,13 +4400,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
if (err < 0)
return err;
} else {
- sprintf(name, "%s Playback Volume", chname[i]);
+ const char *pfx;
+ if (cfg->line_outs == 1 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ else
+ pfx = chname[i];
+ sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", chname[i]);
+ sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 2,
HDA_INPUT));
@@ -4358,31 +4486,61 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
return 0;
}
+static int alc_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
+ return (pincap & AC_PINCAP_IN) != 0;
+}
+
/* create playback/capture controls for input pins */
-static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
+static int alc_auto_create_input_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ hda_nid_t mixer,
+ hda_nid_t cap1, hda_nid_t cap2)
{
+ struct alc_spec *spec = codec->spec;
struct hda_input_mux *imux = &spec->private_imux[0];
int i, err, idx;
for (i = 0; i < AUTO_PIN_LAST; i++) {
- if (alc880_is_input_pin(cfg->input_pins[i])) {
- idx = alc880_input_pin_idx(cfg->input_pins[i]);
- err = new_analog_input(spec, cfg->input_pins[i],
- auto_pin_cfg_labels[i],
- idx, 0x0b);
- if (err < 0)
- return err;
+ hda_nid_t pin;
+
+ pin = cfg->input_pins[i];
+ if (!alc_is_input_pin(codec, pin))
+ continue;
+
+ if (mixer) {
+ idx = get_connection_index(codec, mixer, pin);
+ if (idx >= 0) {
+ err = new_analog_input(spec, pin,
+ auto_pin_cfg_labels[i],
+ idx, mixer);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if (!cap1)
+ continue;
+ idx = get_connection_index(codec, cap1, pin);
+ if (idx < 0 && cap2)
+ idx = get_connection_index(codec, cap2, pin);
+ if (idx >= 0) {
imux->items[imux->num_items].label =
auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index =
- alc880_input_pin_idx(cfg->input_pins[i]);
+ imux->items[imux->num_items].index = idx;
imux->num_items++;
}
}
return 0;
}
+static int alc880_auto_create_input_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x08, 0x09);
+}
+
static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
unsigned int pin_type)
{
@@ -4448,7 +4606,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
- if (alc880_is_input_pin(nid)) {
+ if (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC880_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -4491,7 +4649,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
"Headphone");
if (err < 0)
return err;
- err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc880_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -4505,12 +4663,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
&dig_nid, 1);
if (err < 0)
continue;
- if (dig_nid > 0x7f) {
- printk(KERN_ERR "alc880_auto: invalid dig_nid "
- "connection 0x%x for NID 0x%x\n", dig_nid,
- spec->autocfg.dig_out_pins[i]);
- continue;
- }
if (!i)
spec->multiout.dig_out_nid = dig_nid;
else {
@@ -4547,8 +4699,42 @@ static void alc880_auto_init(struct hda_codec *codec)
alc_inithook(codec);
}
-static void set_capture_mixer(struct alc_spec *spec)
+/* check the ADC/MUX contains all input pins; some ADC/MUX contains only
+ * one of two digital mic pins, e.g. on ALC272
+ */
+static void fixup_automic_adc(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t cap = spec->capsrc_nids ?
+ spec->capsrc_nids[i] : spec->adc_nids[i];
+ int iidx, eidx;
+
+ iidx = get_connection_index(codec, cap, spec->int_mic.pin);
+ if (iidx < 0)
+ continue;
+ eidx = get_connection_index(codec, cap, spec->ext_mic.pin);
+ if (eidx < 0)
+ continue;
+ spec->int_mic.mux_idx = iidx;
+ spec->ext_mic.mux_idx = eidx;
+ if (spec->capsrc_nids)
+ spec->capsrc_nids += i;
+ spec->adc_nids += i;
+ spec->num_adc_nids = 1;
+ return;
+ }
+ snd_printd(KERN_INFO "hda_codec: %s: "
+ "No ADC/MUX containing both 0x%x and 0x%x pins\n",
+ codec->chip_name, spec->int_mic.pin, spec->ext_mic.pin);
+ spec->auto_mic = 0; /* disable auto-mic to be sure */
+}
+
+static void set_capture_mixer(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
static struct snd_kcontrol_new *caps[2][3] = {
{ alc_capture_mixer_nosrc1,
alc_capture_mixer_nosrc2,
@@ -4559,7 +4745,10 @@ static void set_capture_mixer(struct alc_spec *spec)
};
if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
int mux;
- if (spec->input_mux && spec->input_mux->num_items > 1)
+ if (spec->auto_mic) {
+ mux = 0;
+ fixup_automic_adc(codec);
+ } else if (spec->input_mux && spec->input_mux->num_items > 1)
mux = 1;
else
mux = 0;
@@ -4590,8 +4779,8 @@ static int patch_alc880(struct hda_codec *codec)
alc880_models,
alc880_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC880_AUTO;
}
@@ -4616,7 +4805,7 @@ static int patch_alc880(struct hda_codec *codec)
}
if (board_config != ALC880_AUTO)
- setup_preset(spec, &alc880_presets[board_config]);
+ setup_preset(codec, &alc880_presets[board_config]);
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
@@ -4629,7 +4818,7 @@ static int patch_alc880(struct hda_codec *codec)
/* check whether NID 0x07 is valid */
unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
/* get type */
- wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ wcap = get_wcaps_type(wcap);
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc880_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
@@ -4638,7 +4827,7 @@ static int patch_alc880(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
}
}
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x0c;
@@ -5830,7 +6019,14 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = cfg->line_out_pins[0];
if (nid) {
- err = alc260_add_playback_controls(spec, nid, "Front", &vols);
+ const char *pfx;
+ if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
+ pfx = "Master";
+ else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ else
+ pfx = "Front";
+ err = alc260_add_playback_controls(spec, nid, pfx, &vols);
if (err < 0)
return err;
}
@@ -5853,39 +6049,10 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
}
/* create playback/capture controls for input pins */
-static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc260_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- struct hda_input_mux *imux = &spec->private_imux[0];
- int i, err, idx;
-
- for (i = 0; i < AUTO_PIN_LAST; i++) {
- if (cfg->input_pins[i] >= 0x12) {
- idx = cfg->input_pins[i] - 0x12;
- err = new_analog_input(spec, cfg->input_pins[i],
- auto_pin_cfg_labels[i], idx,
- 0x07);
- if (err < 0)
- return err;
- imux->items[imux->num_items].label =
- auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index = idx;
- imux->num_items++;
- }
- if (cfg->input_pins[i] >= 0x0f && cfg->input_pins[i] <= 0x10){
- idx = cfg->input_pins[i] - 0x09;
- err = new_analog_input(spec, cfg->input_pins[i],
- auto_pin_cfg_labels[i], idx,
- 0x07);
- if (err < 0)
- return err;
- imux->items[imux->num_items].label =
- auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index = idx;
- imux->num_items++;
- }
- }
- return 0;
+ return alc_auto_create_input_ctls(codec, cfg, 0x07, 0x04, 0x05);
}
static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -5999,7 +6166,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
return err;
if (!spec->kctls.list)
return 0; /* can't find valid BIOS pin config */
- err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc260_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -6234,8 +6401,7 @@ static int patch_alc260(struct hda_codec *codec)
alc260_models,
alc260_cfg_tbl);
if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n",
+ snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
codec->chip_name);
board_config = ALC260_AUTO;
}
@@ -6261,7 +6427,7 @@ static int patch_alc260(struct hda_codec *codec)
}
if (board_config != ALC260_AUTO)
- setup_preset(spec, &alc260_presets[board_config]);
+ setup_preset(codec, &alc260_presets[board_config]);
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
@@ -6272,7 +6438,7 @@ static int patch_alc260(struct hda_codec *codec)
if (!spec->adc_nids && spec->input_mux) {
/* check whether NID 0x04 is valid */
unsigned int wcap = get_wcaps(codec, 0x04);
- wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ wcap = get_wcaps_type(wcap);
/* get type */
if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc260_adc_nids_alt;
@@ -6282,7 +6448,7 @@ static int patch_alc260(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
}
}
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x08;
@@ -6301,7 +6467,7 @@ static int patch_alc260(struct hda_codec *codec)
/*
- * ALC882 support
+ * ALC882/883/885/888/889 support
*
* ALC882 is almost identical with ALC880 but has cleaner and more flexible
* configuration. Each pin widget can choose any input DACs and a mixer.
@@ -6313,22 +6479,35 @@ static int patch_alc260(struct hda_codec *codec)
*/
#define ALC882_DIGOUT_NID 0x06
#define ALC882_DIGIN_NID 0x0a
+#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
+#define ALC883_DIGIN_NID ALC882_DIGIN_NID
+#define ALC1200_DIGOUT_NID 0x10
+
static struct hda_channel_mode alc882_ch_modes[1] = {
{ 8, NULL }
};
+/* DACs */
static hda_nid_t alc882_dac_nids[4] = {
/* front, rear, clfe, rear_surr */
0x02, 0x03, 0x04, 0x05
};
+#define alc883_dac_nids alc882_dac_nids
-/* identical with ALC880 */
+/* ADCs */
#define alc882_adc_nids alc880_adc_nids
#define alc882_adc_nids_alt alc880_adc_nids_alt
+#define alc883_adc_nids alc882_adc_nids_alt
+static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
+static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
+#define alc889_adc_nids alc880_adc_nids
static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+#define alc883_capsrc_nids alc882_capsrc_nids_alt
+static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
+#define alc889_capsrc_nids alc882_capsrc_nids
/* input MUX */
/* FIXME: should be a matrix-type input source selection */
@@ -6343,6 +6522,17 @@ static struct hda_input_mux alc882_capture_source = {
},
};
+#define alc883_capture_source alc882_capture_source
+
+static struct hda_input_mux alc889_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Mic", 0x3 },
+ { "Line", 0x2 },
+ },
+};
+
static struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
@@ -6352,6 +6542,77 @@ static struct hda_input_mux mb5_capture_source = {
},
};
+static struct hda_input_mux alc883_3stack_6ch_intel = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x1 },
+ { "Front Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "iMic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Int Mic", 0x1 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_sky_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_asus_eee1601_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ },
+};
+
+static struct hda_input_mux alc889A_mb31_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ /* Front Mic (0x01) unused */
+ { "Line", 0x2 },
+ /* Line 2 (0x03) unused */
+ /* CD (0x04) unsused? */
+ },
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
/*
* 2ch mode
*/
@@ -6364,6 +6625,18 @@ static struct hda_verb alc882_3ST_ch2_init[] = {
};
/*
+ * 4ch mode
+ */
+static struct hda_verb alc882_3ST_ch4_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
* 6ch mode
*/
static struct hda_verb alc882_3ST_ch6_init[] = {
@@ -6376,11 +6649,60 @@ static struct hda_verb alc882_3ST_ch6_init[] = {
{ } /* end */
};
-static struct hda_channel_mode alc882_3ST_6ch_modes[2] = {
+static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
{ 2, alc882_3ST_ch2_init },
+ { 4, alc882_3ST_ch4_init },
{ 6, alc882_3ST_ch6_init },
};
+#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_clevo_init[] = {
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
+ { 2, alc883_3ST_ch2_clevo_init },
+ { 4, alc883_3ST_ch4_clevo_init },
+ { 6, alc883_3ST_ch6_clevo_init },
+};
+
+
/*
* 6ch mode
*/
@@ -6468,6 +6790,189 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
{ 6, alc885_mb5_ch6_init },
};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_4ST_ch2_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_4ST_ch4_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_4ST_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_4ST_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
+ { 2, alc883_4ST_ch2_init },
+ { 4, alc883_4ST_ch4_init },
+ { 6, alc883_4ST_ch6_init },
+ { 8, alc883_4ST_ch8_init },
+};
+
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+ { 2, alc883_3ST_ch2_intel_init },
+ { 4, alc883_3ST_ch4_intel_init },
+ { 6, alc883_3ST_ch6_intel_init },
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc889_ch2_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc889_ch6_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc889_ch8_intel_init[] = {
+ { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc889_8ch_intel_modes[3] = {
+ { 2, alc889_ch2_intel_init },
+ { 6, alc889_ch6_intel_init },
+ { 8, alc889_ch8_intel_init },
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_sixstack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_sixstack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_sixstack_modes[2] = {
+ { 6, alc883_sixstack_ch6_init },
+ { 8, alc883_sixstack_ch8_init },
+};
+
+
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
@@ -6604,7 +7109,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = {
{ } /* end */
};
-static struct hda_verb alc882_init_verbs[] = {
+static struct hda_verb alc882_base_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -6622,6 +7127,13 @@ static struct hda_verb alc882_init_verbs[] = {
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute analog input loopbacks */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -6656,11 +7168,6 @@ static struct hda_verb alc882_init_verbs[] = {
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
@@ -6671,9 +7178,6 @@ static struct hda_verb alc882_init_verbs[] = {
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* ADC1: mute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
/* ADC2: mute amp left and right */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -6684,6 +7188,18 @@ static struct hda_verb alc882_init_verbs[] = {
{ }
};
+static struct hda_verb alc882_adc1_init_verbs[] = {
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { }
+};
+
static struct hda_verb alc882_eapd_verbs[] = {
/* change to EAPD mode */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
@@ -6691,6 +7207,110 @@ static struct hda_verb alc882_eapd_verbs[] = {
{ }
};
+static struct hda_verb alc889_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static struct hda_verb alc_hp15_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_verb alc885_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* CLFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Side mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* mute analog input loopbacks */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+ /* Front HP Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Rear Pin: output 1 (0x0d) */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* Mixer elements: 0x18, , 0x1a, 0x1b */
+ /* Input mixer1 */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ { }
+};
+
+static struct hda_verb alc885_init_input_verbs[] = {
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ { }
+};
+
+
+/* Unmute Selector 24h and set the default input to front mic */
+static struct hda_verb alc889_init_input_verbs[] = {
+ {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ { }
+};
+
+
+#define alc883_init_verbs alc882_base_init_verbs
+
/* Mac Pro test */
static struct snd_kcontrol_new alc882_macpro_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -6898,23 +7518,21 @@ static struct hda_verb alc885_imac24_init_verbs[] = {
};
/* Toggle speaker-output according to the hp-jack state */
-static void alc885_imac24_automute_init_hook(struct hda_codec *codec)
+static void alc885_imac24_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x18;
spec->autocfg.speaker_pins[1] = 0x1a;
- alc_automute_amp(codec);
}
-static void alc885_mbp3_init_hook(struct hda_codec *codec)
+static void alc885_mbp3_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_amp(codec);
}
@@ -6942,13 +7560,12 @@ static void alc882_targa_automute(struct hda_codec *codec)
spec->jack_present ? 1 : 3);
}
-static void alc882_targa_init_hook(struct hda_codec *codec)
+static void alc882_targa_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- alc882_targa_automute(codec);
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -7036,18 +7653,16 @@ static void alc885_macpro_init_hook(struct hda_codec *codec)
static void alc885_imac24_init_hook(struct hda_codec *codec)
{
alc885_macpro_init_hook(codec);
- alc885_imac24_automute_init_hook(codec);
+ alc_automute_amp(codec);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
-static struct hda_verb alc882_auto_init_verbs[] = {
+static struct hda_verb alc883_auto_init_verbs[] = {
/*
* Unmute ADC0-2 and set the default input to mic-in
*/
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -7088,11 +7703,6 @@ static struct hda_verb alc882_auto_init_verbs[] = {
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
@@ -7107,821 +7717,6 @@ static struct hda_verb alc882_auto_init_verbs[] = {
{ }
};
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc882_loopbacks alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc882_pcm_analog_playback alc880_pcm_analog_playback
-#define alc882_pcm_analog_capture alc880_pcm_analog_capture
-#define alc882_pcm_digital_playback alc880_pcm_digital_playback
-#define alc882_pcm_digital_capture alc880_pcm_digital_capture
-
-/*
- * configuration and preset
- */
-static const char *alc882_models[ALC882_MODEL_LAST] = {
- [ALC882_3ST_DIG] = "3stack-dig",
- [ALC882_6ST_DIG] = "6stack-dig",
- [ALC882_ARIMA] = "arima",
- [ALC882_W2JC] = "w2jc",
- [ALC882_TARGA] = "targa",
- [ALC882_ASUS_A7J] = "asus-a7j",
- [ALC882_ASUS_A7M] = "asus-a7m",
- [ALC885_MACPRO] = "macpro",
- [ALC885_MB5] = "mb5",
- [ALC885_MBP3] = "mbp3",
- [ALC885_IMAC24] = "imac24",
- [ALC882_AUTO] = "auto",
-};
-
-static struct snd_pci_quirk alc882_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
- SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
- SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
- SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
- SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
- SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
- SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
- {}
-};
-
-static struct alc_config_preset alc882_presets[] = {
- [ALC882_3ST_DIG] = {
- .mixers = { alc882_base_mixer },
- .init_verbs = { alc882_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_6ST_DIG] = {
- .mixers = { alc882_base_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
- .channel_mode = alc882_sixstack_modes,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_ARIMA] = {
- .mixers = { alc882_base_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
- .channel_mode = alc882_sixstack_modes,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_W2JC] = {
- .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- },
- [ALC885_MBP3] = {
- .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mbp3_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = 2,
- .dac_nids = alc882_dac_nids,
- .hp_nid = 0x04,
- .channel_mode = alc885_mbp_4ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
- .input_mux = &alc882_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc885_mbp3_init_hook,
- },
- [ALC885_MB5] = {
- .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mb5_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mb5_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
- .input_mux = &mb5_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- },
- [ALC885_MACPRO] = {
- .mixers = { alc882_macpro_mixer },
- .init_verbs = { alc882_macpro_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .input_mux = &alc882_capture_source,
- .init_hook = alc885_macpro_init_hook,
- },
- [ALC885_IMAC24] = {
- .mixers = { alc885_imac24_mixer },
- .init_verbs = { alc885_imac24_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
- .channel_mode = alc882_ch_modes,
- .input_mux = &alc882_capture_source,
- .unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc885_imac24_init_hook,
- },
- [ALC882_TARGA] = {
- .mixers = { alc882_targa_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs,
- alc882_targa_verbs},
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
- .adc_nids = alc882_adc_nids,
- .capsrc_nids = alc882_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
- .channel_mode = alc882_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- .unsol_event = alc882_targa_unsol_event,
- .init_hook = alc882_targa_init_hook,
- },
- [ALC882_ASUS_A7J] = {
- .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
- .adc_nids = alc882_adc_nids,
- .capsrc_nids = alc882_capsrc_nids,
- .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
- .channel_mode = alc882_3ST_6ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
- [ALC882_ASUS_A7M] = {
- .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
- .init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
- alc880_gpio1_init_verbs,
- alc882_asus_a7m_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc882_capture_source,
- },
-};
-
-
-/*
- * Pin config fixes
- */
-enum {
- PINFIX_ABIT_AW9D_MAX
-};
-
-static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
- { 0x15, 0x01080104 }, /* side */
- { 0x16, 0x01011012 }, /* rear */
- { 0x17, 0x01016011 }, /* clfe */
- { }
-};
-
-static const struct alc_pincfg *alc882_pin_fixes[] = {
- [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
-};
-
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
- SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
- {}
-};
-
-/*
- * BIOS auto configuration
- */
-static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
- hda_nid_t nid, int pin_type,
- int dac_idx)
-{
- /* set as output */
- struct alc_spec *spec = codec->spec;
- int idx;
-
- alc_set_pin_output(codec, nid, pin_type);
- if (spec->multiout.dac_nids[dac_idx] == 0x25)
- idx = 4;
- else
- idx = spec->multiout.dac_nids[dac_idx] - 2;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
-
-}
-
-static void alc882_auto_init_multi_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i <= HDA_SIDE; i++) {
- hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
- if (nid)
- alc882_auto_set_output_and_unmute(codec, nid, pin_type,
- i);
- }
-}
-
-static void alc882_auto_init_hp_out(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
-
- pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
- /* use dac 0 */
- alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
- pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
-}
-
-#define alc882_is_input_pin(nid) alc880_is_input_pin(nid)
-#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID
-
-static void alc882_auto_init_analog_input(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < AUTO_PIN_LAST; i++) {
- hda_nid_t nid = spec->autocfg.input_pins[i];
- if (!nid)
- continue;
- alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
- if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- }
-}
-
-static void alc882_auto_init_input_src(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int c;
-
- for (c = 0; c < spec->num_adc_nids; c++) {
- hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
- hda_nid_t nid = spec->capsrc_nids[c];
- unsigned int mux_idx;
- const struct hda_input_mux *imux;
- int conns, mute, idx, item;
-
- conns = snd_hda_get_connections(codec, nid, conn_list,
- ARRAY_SIZE(conn_list));
- if (conns < 0)
- continue;
- mux_idx = c >= spec->num_mux_defs ? 0 : c;
- imux = &spec->input_mux[mux_idx];
- for (idx = 0; idx < conns; idx++) {
- /* if the current connection is the selected one,
- * unmute it as default - otherwise mute it
- */
- mute = AMP_IN_MUTE(idx);
- for (item = 0; item < imux->num_items; item++) {
- if (imux->items[item].index == idx) {
- if (spec->cur_mux[c] == item)
- mute = AMP_IN_UNMUTE(idx);
- break;
- }
- }
- /* check if we have a selector or mixer
- * we could check for the widget type instead, but
- * just check for Amp-In presence (in case of mixer
- * without amp-in there is something wrong, this
- * function shouldn't be used or capsrc nid is wrong)
- */
- if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- mute);
- else if (mute != AMP_IN_MUTE(idx))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL,
- idx);
- }
- }
-}
-
-/* add mic boosts if needed */
-static int alc_auto_add_mic_boost(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err;
- hda_nid_t nid;
-
- nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
- if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Mic Boost",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- }
- nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
- if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Front Mic Boost",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
- if (err < 0)
- return err;
- }
- return 0;
-}
-
-/* almost identical with ALC880 parser... */
-static int alc882_parse_auto_config(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- int err = alc880_parse_auto_config(codec);
-
- if (err < 0)
- return err;
- else if (!err)
- return 0; /* no config found */
-
- err = alc_auto_add_mic_boost(codec);
- if (err < 0)
- return err;
-
- /* hack - override the init verbs */
- spec->init_verbs[0] = alc882_auto_init_verbs;
-
- return 1; /* config found */
-}
-
-/* additional initialization for auto-configuration model */
-static void alc882_auto_init(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- alc882_auto_init_multi_out(codec);
- alc882_auto_init_hp_out(codec);
- alc882_auto_init_analog_input(codec);
- alc882_auto_init_input_src(codec);
- if (spec->unsol_event)
- alc_inithook(codec);
-}
-
-static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */
-
-static int patch_alc882(struct hda_codec *codec)
-{
- struct alc_spec *spec;
- int err, board_config;
-
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
- alc882_models,
- alc882_cfg_tbl);
-
- if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
- /* Pick up systems that don't supply PCI SSID */
- switch (codec->subsystem_id) {
- case 0x106b0c00: /* Mac Pro */
- board_config = ALC885_MACPRO;
- break;
- case 0x106b1000: /* iMac 24 */
- case 0x106b2800: /* AppleTV */
- case 0x106b3e00: /* iMac 24 Aluminium */
- board_config = ALC885_IMAC24;
- break;
- case 0x106b00a0: /* MacBookPro3,1 - Another revision */
- case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
- case 0x106b00a4: /* MacbookPro4,1 */
- case 0x106b2c00: /* Macbook Pro rev3 */
- /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */
- case 0x106b3800: /* MacbookPro4,1 - latter revision */
- board_config = ALC885_MBP3;
- break;
- case 0x106b3f00: /* Macbook 5,1 */
- case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense
- * seems not working, so apparently
- * no perfect solution yet
- */
- board_config = ALC885_MB5;
- break;
- default:
- /* ALC889A is handled better as ALC888-compatible */
- if (codec->revision_id == 0x100101 ||
- codec->revision_id == 0x100103) {
- alc_free(codec);
- return patch_alc883(codec);
- }
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n",
- codec->chip_name);
- board_config = ALC882_AUTO;
- }
- }
-
- alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
-
- if (board_config == ALC882_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc882_parse_auto_config(codec);
- if (err < 0) {
- alc_free(codec);
- return err;
- } else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC882_3ST_DIG;
- }
- }
-
- err = snd_hda_attach_beep_device(codec, 0x1);
- if (err < 0) {
- alc_free(codec);
- return err;
- }
-
- if (board_config != ALC882_AUTO)
- setup_preset(spec, &alc882_presets[board_config]);
-
- spec->stream_analog_playback = &alc882_pcm_analog_playback;
- spec->stream_analog_capture = &alc882_pcm_analog_capture;
- /* FIXME: setup DAC5 */
- /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
- spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
-
- spec->stream_digital_playback = &alc882_pcm_digital_playback;
- spec->stream_digital_capture = &alc882_pcm_digital_capture;
-
- if (!spec->adc_nids && spec->input_mux) {
- /* check whether NID 0x07 is valid */
- unsigned int wcap = get_wcaps(codec, 0x07);
- /* get type */
- wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
- if (wcap != AC_WID_AUD_IN) {
- spec->adc_nids = alc882_adc_nids_alt;
- spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
- spec->capsrc_nids = alc882_capsrc_nids_alt;
- } else {
- spec->adc_nids = alc882_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
- spec->capsrc_nids = alc882_capsrc_nids;
- }
- }
- set_capture_mixer(spec);
- set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
- spec->vmaster_nid = 0x0c;
-
- codec->patch_ops = alc_patch_ops;
- if (board_config == ALC882_AUTO)
- spec->init_hook = alc882_auto_init;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc882_loopbacks;
-#endif
- codec->proc_widget_hook = print_realtek_coef;
-
- return 0;
-}
-
-/*
- * ALC883 support
- *
- * ALC883 is almost identical with ALC880 but has cleaner and more flexible
- * configuration. Each pin widget can choose any input DACs and a mixer.
- * Each ADC is connected from a mixer of all inputs. This makes possible
- * 6-channel independent captures.
- *
- * In addition, an independent DAC for the multi-playback (not used in this
- * driver yet).
- */
-#define ALC883_DIGOUT_NID 0x06
-#define ALC883_DIGIN_NID 0x0a
-
-#define ALC1200_DIGOUT_NID 0x10
-
-static hda_nid_t alc883_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x03, 0x04, 0x05
-};
-
-static hda_nid_t alc883_adc_nids[2] = {
- /* ADC1-2 */
- 0x08, 0x09,
-};
-
-static hda_nid_t alc883_adc_nids_alt[1] = {
- /* ADC1 */
- 0x08,
-};
-
-static hda_nid_t alc883_adc_nids_rev[2] = {
- /* ADC2-1 */
- 0x09, 0x08
-};
-
-#define alc889_adc_nids alc880_adc_nids
-
-static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
-
-static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
-
-#define alc889_capsrc_nids alc882_capsrc_nids
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-
-static struct hda_input_mux alc883_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static struct hda_input_mux alc883_3stack_6ch_intel = {
- .num_items = 4,
- .items = {
- { "Mic", 0x1 },
- { "Front Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static struct hda_input_mux alc883_lenovo_101e_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x2 },
- },
-};
-
-static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "iMic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Int Mic", 0x1 },
- },
-};
-
-static struct hda_input_mux alc883_lenovo_sky_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x4 },
- },
-};
-
-static struct hda_input_mux alc883_asus_eee1601_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- },
-};
-
-static struct hda_input_mux alc889A_mb31_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- /* Front Mic (0x01) unused */
- { "Line", 0x2 },
- /* Line 2 (0x03) unused */
- /* CD (0x04) unsused? */
- },
-};
-
-/*
- * 2ch mode
- */
-static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
- { 2, NULL }
-};
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_3ST_ch2_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_3ST_ch4_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_3ST_ch6_init[] = {
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static struct hda_channel_mode alc883_3ST_6ch_modes[3] = {
- { 2, alc883_3ST_ch2_init },
- { 4, alc883_3ST_ch4_init },
- { 6, alc883_3ST_ch6_init },
-};
-
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_4ST_ch2_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_4ST_ch4_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_4ST_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static struct hda_verb alc883_4ST_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
- { 2, alc883_4ST_ch2_init },
- { 4, alc883_4ST_ch4_init },
- { 6, alc883_4ST_ch6_init },
- { 8, alc883_4ST_ch8_init },
-};
-
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_3ST_ch2_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_3ST_ch4_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_3ST_ch6_intel_init[] = {
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { } /* end */
-};
-
-static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
- { 2, alc883_3ST_ch2_intel_init },
- { 4, alc883_3ST_ch4_intel_init },
- { 6, alc883_3ST_ch6_intel_init },
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_sixstack_ch6_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-/*
- * 8ch mode
- */
-static struct hda_verb alc883_sixstack_ch8_init[] = {
- { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { } /* end */
-};
-
-static struct hda_channel_mode alc883_sixstack_modes[2] = {
- { 6, alc883_sixstack_ch6_init },
- { 8, alc883_sixstack_ch8_init },
-};
-
/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
static struct hda_verb alc889A_mb31_ch2_init[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
@@ -7972,34 +7767,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = {
{ }
};
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-
-static struct snd_kcontrol_new alc883_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
+#define alc883_base_mixer alc882_base_mixer
static struct snd_kcontrol_new alc883_mitac_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -8110,6 +7878,30 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x1b, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -8350,93 +8142,14 @@ static struct snd_kcontrol_new alc883_chmode_mixer[] = {
{ } /* end */
};
-static struct hda_verb alc883_init_verbs[] = {
- /* ADC1: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC2: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* CLFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Side mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- /* mute analog input loopbacks */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Rear Pin: output 1 (0x0d) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* Side Pin: output 3 (0x0f) */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- { }
-};
-
/* toggle speaker-output according to the hp-jack state */
-static void alc883_mitac_init_hook(struct hda_codec *codec)
+static void alc883_mitac_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- alc_automute_amp(codec);
}
/* auto-toggle front mic */
@@ -8468,6 +8181,22 @@ static struct hda_verb alc883_mitac_verbs[] = {
{ } /* end */
};
+static struct hda_verb alc883_clevo_m540r_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Int speaker */
+ /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/
+
+ /* enable unsolicited event */
+ /*
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+ */
+
+ { } /* end */
+};
+
static struct hda_verb alc883_clevo_m720_verbs[] = {
/* HP */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -8591,7 +8320,7 @@ static struct hda_verb alc883_vaiott_verbs[] = {
{ } /* end */
};
-static void alc888_3st_hp_init_hook(struct hda_codec *codec)
+static void alc888_3st_hp_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -8599,7 +8328,6 @@ static void alc888_3st_hp_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x16;
spec->autocfg.speaker_pins[2] = 0x18;
- alc_automute_amp(codec);
}
static struct hda_verb alc888_3st_hp_verbs[] = {
@@ -8696,13 +8424,12 @@ static struct hda_verb alc883_medion_md2_verbs[] = {
};
/* toggle speaker-output according to the hp-jack state */
-static void alc883_medion_md2_init_hook(struct hda_codec *codec)
+static void alc883_medion_md2_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_automute_amp(codec);
}
/* toggle speaker-output according to the hp-jack state */
@@ -8719,12 +8446,16 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
-static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
+static void alc883_clevo_m720_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
+}
+
+static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
+{
alc_automute_amp(codec);
alc883_clevo_m720_mic_automute(codec);
}
@@ -8743,22 +8474,20 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
}
/* toggle speaker-output according to the hp-jack state */
-static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec)
+static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_automute_amp(codec);
}
-static void alc883_haier_w66_init_hook(struct hda_codec *codec)
+static void alc883_haier_w66_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_amp(codec);
}
static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
@@ -8797,14 +8526,13 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
}
/* toggle speaker-output according to the hp-jack state */
-static void alc883_acer_aspire_init_hook(struct hda_codec *codec)
+static void alc883_acer_aspire_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x15;
spec->autocfg.speaker_pins[1] = 0x16;
- alc_automute_amp(codec);
}
static struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -8825,7 +8553,14 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
{ }
};
-static void alc888_6st_dell_init_hook(struct hda_codec *codec)
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static void alc888_6st_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -8834,10 +8569,9 @@ static void alc888_6st_dell_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
- alc_automute_amp(codec);
}
-static void alc888_lenovo_sky_init_hook(struct hda_codec *codec)
+static void alc888_lenovo_sky_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -8847,82 +8581,17 @@ static void alc888_lenovo_sky_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[2] = 0x16;
spec->autocfg.speaker_pins[3] = 0x17;
spec->autocfg.speaker_pins[4] = 0x1a;
- alc_automute_amp(codec);
}
-static void alc883_vaiott_init_hook(struct hda_codec *codec)
+static void alc883_vaiott_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x17;
- alc_automute_amp(codec);
}
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static struct hda_verb alc883_auto_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for
- * front panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-
- /*
- * Set up output mixers (0x0c - 0x0f)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- /* Input mixer2 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
- { }
-};
-
static struct hda_verb alc888_asus_m90v_verbs[] = {
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -8933,19 +8602,7 @@ static struct hda_verb alc888_asus_m90v_verbs[] = {
{ } /* end */
};
-static void alc883_nb_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-}
-
-static void alc883_M90V_init_hook(struct hda_codec *codec)
+static void alc883_mode2_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -8953,26 +8610,11 @@ static void alc883_M90V_init_hook(struct hda_codec *codec)
spec->autocfg.speaker_pins[0] = 0x14;
spec->autocfg.speaker_pins[1] = 0x15;
spec->autocfg.speaker_pins[2] = 0x16;
- alc_automute_pin(codec);
-}
-
-static void alc883_mode2_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_MIC_EVENT:
- alc883_nb_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
-}
-
-static void alc883_mode2_inithook(struct hda_codec *codec)
-{
- alc883_M90V_init_hook(codec);
- alc883_nb_mic_automute(codec);
+ spec->ext_mic.pin = 0x18;
+ spec->int_mic.pin = 0x19;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
}
static struct hda_verb alc888_asus_eee1601_verbs[] = {
@@ -9033,25 +8675,44 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
alc889A_mb31_automute(codec);
}
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc883_loopbacks alc880_loopbacks
+#define alc882_loopbacks alc880_loopbacks
#endif
/* pcm configuration: identical with ALC880 */
-#define alc883_pcm_analog_playback alc880_pcm_analog_playback
-#define alc883_pcm_analog_capture alc880_pcm_analog_capture
-#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
-#define alc883_pcm_digital_playback alc880_pcm_digital_playback
-#define alc883_pcm_digital_capture alc880_pcm_digital_capture
+#define alc882_pcm_analog_playback alc880_pcm_analog_playback
+#define alc882_pcm_analog_capture alc880_pcm_analog_capture
+#define alc882_pcm_digital_playback alc880_pcm_digital_playback
+#define alc882_pcm_digital_capture alc880_pcm_digital_capture
+
+static hda_nid_t alc883_slave_dig_outs[] = {
+ ALC1200_DIGOUT_NID, 0,
+};
+
+static hda_nid_t alc1200_slave_dig_outs[] = {
+ ALC883_DIGOUT_NID, 0,
+};
/*
* configuration and preset
*/
-static const char *alc883_models[ALC883_MODEL_LAST] = {
- [ALC883_3ST_2ch_DIG] = "3stack-dig",
+static const char *alc882_models[ALC882_MODEL_LAST] = {
+ [ALC882_3ST_DIG] = "3stack-dig",
+ [ALC882_6ST_DIG] = "6stack-dig",
+ [ALC882_ARIMA] = "arima",
+ [ALC882_W2JC] = "w2jc",
+ [ALC882_TARGA] = "targa",
+ [ALC882_ASUS_A7J] = "asus-a7j",
+ [ALC882_ASUS_A7M] = "asus-a7m",
+ [ALC885_MACPRO] = "macpro",
+ [ALC885_MB5] = "mb5",
+ [ALC885_MBP3] = "mbp3",
+ [ALC885_IMAC24] = "imac24",
+ [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig",
[ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
[ALC883_3ST_6ch] = "3stack-6ch",
- [ALC883_6ST_DIG] = "6stack-dig",
+ [ALC883_6ST_DIG] = "alc883-6stack-dig",
[ALC883_TARGA_DIG] = "targa-dig",
[ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
[ALC883_TARGA_8ch_DIG] = "targa-8ch-dig",
@@ -9060,6 +8721,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
[ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g",
[ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
+ [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
@@ -9071,18 +8733,22 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC888_3ST_HP] = "3stack-hp",
[ALC888_6ST_DELL] = "6stack-dell",
[ALC883_MITAC] = "mitac",
+ [ALC883_CLEVO_M540R] = "clevo-m540r",
[ALC883_CLEVO_M720] = "clevo-m720",
[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
[ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
+ [ALC889A_INTEL] = "intel-alc889a",
+ [ALC889_INTEL] = "intel-x58",
[ALC1200_ASUS_P5Q] = "asus-p5q",
[ALC889A_MB31] = "mb31",
[ALC883_SONY_VAIO_TT] = "sony-vaio-tt",
- [ALC883_AUTO] = "auto",
+ [ALC882_AUTO] = "auto",
};
-static struct snd_pci_quirk alc883_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
+static struct snd_pci_quirk alc882_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
+
SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
@@ -9097,40 +8763,56 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
ALC888_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
ALC888_ACER_ASPIRE_8930G),
- SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO),
- SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
+ SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
+ SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC888_ACER_ASPIRE_6530G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
ALC888_ACER_ASPIRE_6530G),
+ SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+ ALC888_ACER_ASPIRE_7730G),
/* default Acer -- disabled as it causes more problems.
* model=auto should work fine now
*/
/* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
+
SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
+
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
+
+ SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
+ SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+ SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
+
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
- SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
- SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
+
SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
@@ -9139,6 +8821,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
@@ -9152,11 +8835,15 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
+
SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
+ /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
ALC883_FUJITSU_PI2515),
@@ -9171,24 +8858,186 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+
SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
- SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
- SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
- {}
-};
-static hda_nid_t alc883_slave_dig_outs[] = {
- ALC1200_DIGOUT_NID, 0,
+ {}
};
-static hda_nid_t alc1200_slave_dig_outs[] = {
- ALC883_DIGOUT_NID, 0,
+/* codec SSID table for Intel Mac */
+static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
+ SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
+ SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
+ SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
+ SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+ /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
+ * no perfect solution yet
+ */
+ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+ {} /* terminator */
};
-static struct alc_config_preset alc883_presets[] = {
+static struct alc_config_preset alc882_presets[] = {
+ [ALC882_3ST_DIG] = {
+ .mixers = { alc882_base_mixer },
+ .init_verbs = { alc882_base_init_verbs,
+ alc882_adc1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_6ST_DIG] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs,
+ alc882_adc1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ARIMA] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_W2JC] = {
+ .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs, alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ },
+ [ALC885_MBP3] = {
+ .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mbp3_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = 2,
+ .dac_nids = alc882_dac_nids,
+ .hp_nid = 0x04,
+ .channel_mode = alc885_mbp_4ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .setup = alc885_mbp3_setup,
+ .init_hook = alc_automute_amp,
+ },
+ [ALC885_MB5] = {
+ .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mb5_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mb5_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
+ .input_mux = &mb5_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ },
+ [ALC885_MACPRO] = {
+ .mixers = { alc882_macpro_mixer },
+ .init_verbs = { alc882_macpro_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .init_hook = alc885_macpro_init_hook,
+ },
+ [ALC885_IMAC24] = {
+ .mixers = { alc885_imac24_mixer },
+ .init_verbs = { alc885_imac24_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .setup = alc885_imac24_setup,
+ .init_hook = alc885_imac24_init_hook,
+ },
+ [ALC882_TARGA] = {
+ .mixers = { alc882_targa_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc880_gpio3_init_verbs, alc882_targa_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc882_targa_unsol_event,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC882_ASUS_A7J] = {
+ .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_asus_a7j_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ASUS_A7M] = {
+ .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+ alc882_eapd_verbs, alc880_gpio1_init_verbs,
+ alc882_asus_a7m_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
[ALC883_3ST_2ch_DIG] = {
.mixers = { alc883_3ST_2ch_mixer },
.init_verbs = { alc883_init_verbs },
@@ -9235,6 +9084,46 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_3stack_6ch_intel,
},
+ [ALC889A_INTEL] = {
+ .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
+ alc_hp15_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+ .channel_mode = alc889_8ch_intel_modes,
+ .capsrc_nids = alc889_capsrc_nids,
+ .input_mux = &alc889_capture_source,
+ .setup = alc889_automute_setup,
+ .init_hook = alc_automute_amp,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .need_dac_fix = 1,
+ },
+ [ALC889_INTEL] = {
+ .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
+ alc889_eapd_verbs, alc_hp15_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .slave_dig_outs = alc883_slave_dig_outs,
+ .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+ .channel_mode = alc889_8ch_intel_modes,
+ .capsrc_nids = alc889_capsrc_nids,
+ .input_mux = &alc889_capture_source,
+ .setup = alc889_automute_setup,
+ .init_hook = alc889_intel_init_hook,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .need_dac_fix = 1,
+ },
[ALC883_6ST_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs },
@@ -9258,7 +9147,8 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
- .init_hook = alc883_targa_init_hook,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_targa_2ch_mixer},
@@ -9273,7 +9163,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
- .init_hook = alc883_targa_init_hook,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
},
[ALC883_TARGA_8ch_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -9291,7 +9182,8 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_targa_unsol_event,
- .init_hook = alc883_targa_init_hook,
+ .setup = alc882_targa_setup,
+ .init_hook = alc882_targa_automute,
},
[ALC883_ACER] = {
.mixers = { alc883_base_mixer },
@@ -9317,7 +9209,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_acer_aspire_init_hook,
+ .setup = alc883_acer_aspire_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_4930G] = {
.mixers = { alc888_base_mixer,
@@ -9337,7 +9230,8 @@ static struct alc_config_preset alc883_presets[] = {
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_acer_aspire_4930g_init_hook,
+ .setup = alc888_acer_aspire_4930g_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_6530G] = {
.mixers = { alc888_acer_aspire_6530_mixer },
@@ -9355,7 +9249,8 @@ static struct alc_config_preset alc883_presets[] = {
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_acer_aspire_6530_sources,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_acer_aspire_6530g_init_hook,
+ .setup = alc888_acer_aspire_6530g_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_ACER_ASPIRE_8930G] = {
.mixers = { alc888_base_mixer,
@@ -9376,7 +9271,28 @@ static struct alc_config_preset alc883_presets[] = {
ARRAY_SIZE(alc889_capture_sources),
.input_mux = alc889_capture_sources,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc889_acer_aspire_8930g_init_hook,
+ .setup = alc889_acer_aspire_8930g_setup,
+ .init_hook = alc_automute_amp,
+ },
+ [ALC888_ACER_ASPIRE_7730G] = {
+ .mixers = { alc883_3ST_6ch_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc888_acer_aspire_7730G_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .const_channel_count = 6,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .setup = alc888_acer_aspire_6530g_setup,
+ .init_hook = alc_automute_amp,
},
[ALC883_MEDION] = {
.mixers = { alc883_fivestack_mixer,
@@ -9401,7 +9317,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_medion_md2_init_hook,
+ .setup = alc883_medion_md2_setup,
+ .init_hook = alc_automute_amp,
},
[ALC883_LAPTOP_EAPD] = {
.mixers = { alc883_base_mixer },
@@ -9412,6 +9329,21 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
+ [ALC883_CLEVO_M540R] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes),
+ .channel_mode = alc883_3ST_6ch_clevo_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ /* This machine has the hardware HP auto-muting, thus
+ * we need no software mute via unsol event
+ */
+ },
[ALC883_CLEVO_M720] = {
.mixers = { alc883_clevo_m720_mixer },
.init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
@@ -9422,6 +9354,7 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_clevo_m720_unsol_event,
+ .setup = alc883_clevo_m720_setup,
.init_hook = alc883_clevo_m720_init_hook,
},
[ALC883_LENOVO_101E_2ch] = {
@@ -9447,7 +9380,8 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_nb0763_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_medion_md2_init_hook,
+ .setup = alc883_medion_md2_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_LENOVO_MS7195_DIG] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9472,7 +9406,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_haier_w66_init_hook,
+ .setup = alc883_haier_w66_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_3ST_HP] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9484,7 +9419,8 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_3st_hp_init_hook,
+ .setup = alc888_3st_hp_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_6ST_DELL] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -9497,7 +9433,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_6st_dell_init_hook,
+ .setup = alc888_6st_dell_setup,
+ .init_hook = alc_automute_amp,
},
[ALC883_MITAC] = {
.mixers = { alc883_mitac_mixer },
@@ -9508,7 +9445,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_mitac_init_hook,
+ .setup = alc883_mitac_setup,
+ .init_hook = alc_automute_amp,
},
[ALC883_FUJITSU_PI2515] = {
.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
@@ -9521,7 +9459,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_fujitsu_pi2515_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_2ch_fujitsu_pi2515_init_hook,
+ .setup = alc883_2ch_fujitsu_pi2515_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_FUJITSU_XA3530] = {
.mixers = { alc888_base_mixer, alc883_chmode_mixer },
@@ -9539,7 +9478,8 @@ static struct alc_config_preset alc883_presets[] = {
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_fujitsu_xa3530_init_hook,
+ .setup = alc888_fujitsu_xa3530_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_LENOVO_SKY] = {
.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
@@ -9552,7 +9492,8 @@ static struct alc_config_preset alc883_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_sky_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc888_lenovo_sky_init_hook,
+ .setup = alc888_lenovo_sky_setup,
+ .init_hook = alc_automute_amp,
},
[ALC888_ASUS_M90V] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9565,8 +9506,9 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_fujitsu_pi2515_capture_source,
- .unsol_event = alc883_mode2_unsol_event,
- .init_hook = alc883_mode2_inithook,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc883_mode2_setup,
+ .init_hook = alc_inithook,
},
[ALC888_ASUS_EEE1601] = {
.mixers = { alc883_asus_eee1601_mixer },
@@ -9619,15 +9561,45 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc883_vaiott_init_hook,
+ .setup = alc883_vaiott_setup,
+ .init_hook = alc_automute_amp,
},
};
/*
+ * Pin config fixes
+ */
+enum {
+ PINFIX_ABIT_AW9D_MAX
+};
+
+static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
+ { 0x15, 0x01080104 }, /* side */
+ { 0x16, 0x01011012 }, /* rear */
+ { 0x17, 0x01016011 }, /* clfe */
+ { }
+};
+
+static const struct alc_pincfg *alc882_pin_fixes[] = {
+ [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+};
+
+static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+ SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
+ {}
+};
+
+/*
* BIOS auto configuration
*/
-static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
+static int alc882_auto_create_input_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x23, 0x22);
+}
+
+static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
int dac_idx)
{
@@ -9644,7 +9616,7 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
}
-static void alc883_auto_init_multi_out(struct hda_codec *codec)
+static void alc882_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
@@ -9653,12 +9625,12 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec)
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
- alc883_auto_set_output_and_unmute(codec, nid, pin_type,
+ alc882_auto_set_output_and_unmute(codec, nid, pin_type,
i);
}
}
-static void alc883_auto_init_hp_out(struct hda_codec *codec)
+static void alc882_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t pin;
@@ -9666,91 +9638,191 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec)
pin = spec->autocfg.hp_pins[0];
if (pin) /* connect to front */
/* use dac 0 */
- alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
pin = spec->autocfg.speaker_pins[0];
if (pin)
- alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
-#define alc883_is_input_pin(nid) alc880_is_input_pin(nid)
-#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID
-
-static void alc883_auto_init_analog_input(struct hda_codec *codec)
+static void alc882_auto_init_analog_input(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
int i;
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
- if (alc883_is_input_pin(nid)) {
- alc_set_input_pin(codec, nid, i);
- if (nid != ALC883_PIN_CD_NID &&
- (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
+ if (!nid)
+ continue;
+ alc_set_input_pin(codec, nid, i);
+ if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+}
+
+static void alc882_auto_init_input_src(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int c;
+
+ for (c = 0; c < spec->num_adc_nids; c++) {
+ hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
+ hda_nid_t nid = spec->capsrc_nids[c];
+ unsigned int mux_idx;
+ const struct hda_input_mux *imux;
+ int conns, mute, idx, item;
+
+ conns = snd_hda_get_connections(codec, nid, conn_list,
+ ARRAY_SIZE(conn_list));
+ if (conns < 0)
+ continue;
+ mux_idx = c >= spec->num_mux_defs ? 0 : c;
+ imux = &spec->input_mux[mux_idx];
+ for (idx = 0; idx < conns; idx++) {
+ /* if the current connection is the selected one,
+ * unmute it as default - otherwise mute it
+ */
+ mute = AMP_IN_MUTE(idx);
+ for (item = 0; item < imux->num_items; item++) {
+ if (imux->items[item].index == idx) {
+ if (spec->cur_mux[c] == item)
+ mute = AMP_IN_UNMUTE(idx);
+ break;
+ }
+ }
+ /* check if we have a selector or mixer
+ * we could check for the widget type instead, but
+ * just check for Amp-In presence (in case of mixer
+ * without amp-in there is something wrong, this
+ * function shouldn't be used or capsrc nid is wrong)
+ */
+ if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
+ mute);
+ else if (mute != AMP_IN_MUTE(idx))
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ idx);
}
}
}
-#define alc883_auto_init_input_src alc882_auto_init_input_src
+/* add mic boosts if needed */
+static int alc_auto_add_mic_boost(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ hda_nid_t nid;
+
+ nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Front Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
/* almost identical with ALC880 parser... */
-static int alc883_parse_auto_config(struct hda_codec *codec)
+static int alc882_parse_auto_config(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- int err = alc880_parse_auto_config(codec);
- struct auto_pin_cfg *cfg = &spec->autocfg;
- int i;
+ static hda_nid_t alc882_ignore[] = { 0x1d, 0 };
+ int i, err;
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc882_ignore);
if (err < 0)
return err;
- else if (!err)
- return 0; /* no config found */
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
- err = alc_auto_add_mic_boost(codec);
+ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_extra_out(spec,
+ spec->autocfg.speaker_pins[0],
+ "Speaker");
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+ "Headphone");
+ if (err < 0)
+ return err;
+ err = alc882_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
- /* hack - override the init verbs */
- spec->init_verbs[0] = alc883_auto_init_verbs;
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
- /* setup input_mux for ALC889 */
- if (codec->vendor_id == 0x10ec0889) {
- /* digital-mic input pin is excluded in alc880_auto_create..()
- * because it's under 0x18
- */
- if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
- cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
- struct hda_input_mux *imux = &spec->private_imux[0];
- for (i = 1; i < 3; i++)
- memcpy(&spec->private_imux[i],
- &spec->private_imux[0],
- sizeof(spec->private_imux[0]));
- imux->items[imux->num_items].label = "Int DMic";
- imux->items[imux->num_items].index = 0x0b;
- imux->num_items++;
- spec->num_mux_defs = 3;
- spec->input_mux = spec->private_imux;
+ /* check multiple SPDIF-out (for recent codecs) */
+ for (i = 0; i < spec->autocfg.dig_outs; i++) {
+ hda_nid_t dig_nid;
+ err = snd_hda_get_connections(codec,
+ spec->autocfg.dig_out_pins[i],
+ &dig_nid, 1);
+ if (err < 0)
+ continue;
+ if (!i)
+ spec->multiout.dig_out_nid = dig_nid;
+ else {
+ spec->multiout.slave_dig_outs = spec->slave_dig_outs;
+ spec->slave_dig_outs[i - 1] = dig_nid;
+ if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ break;
}
}
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = ALC880_DIGIN_NID;
+
+ if (spec->kctls.list)
+ add_mixer(spec, spec->kctls.list);
+
+ add_verb(spec, alc883_auto_init_verbs);
+ /* if ADC 0x07 is available, initialize it, too */
+ if (get_wcaps_type(get_wcaps(codec, 0x07)) == AC_WID_AUD_IN)
+ add_verb(spec, alc882_adc1_init_verbs);
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux[0];
+
+ alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
return 1; /* config found */
}
/* additional initialization for auto-configuration model */
-static void alc883_auto_init(struct hda_codec *codec)
+static void alc882_auto_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- alc883_auto_init_multi_out(codec);
- alc883_auto_init_hp_out(codec);
- alc883_auto_init_analog_input(codec);
- alc883_auto_init_input_src(codec);
+ alc882_auto_init_multi_out(codec);
+ alc882_auto_init_hp_out(codec);
+ alc882_auto_init_analog_input(codec);
+ alc882_auto_init_input_src(codec);
if (spec->unsol_event)
alc_inithook(codec);
}
-static int patch_alc883(struct hda_codec *codec)
+static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
int err, board_config;
@@ -9761,28 +9833,35 @@ static int patch_alc883(struct hda_codec *codec)
codec->spec = spec;
- alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+ switch (codec->vendor_id) {
+ case 0x10ec0882:
+ case 0x10ec0885:
+ break;
+ default:
+ /* ALC883 and variants */
+ alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+ break;
+ }
- board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
- alc883_models,
- alc883_cfg_tbl);
- if (board_config < 0 || board_config >= ALC883_MODEL_LAST) {
- /* Pick up systems that don't supply PCI SSID */
- switch (codec->subsystem_id) {
- case 0x106b3600: /* Macbook 3.1 */
- board_config = ALC889A_MB31;
- break;
- default:
- printk(KERN_INFO
- "hda_codec: Unknown model for %s, trying "
- "auto-probe from BIOS...\n", codec->chip_name);
- board_config = ALC883_AUTO;
- }
+ board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
+ alc882_models,
+ alc882_cfg_tbl);
+
+ if (board_config < 0 || board_config >= ALC882_MODEL_LAST)
+ board_config = snd_hda_check_board_codec_sid_config(codec,
+ ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
+
+ if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
+ board_config = ALC882_AUTO;
}
- if (board_config == ALC883_AUTO) {
+ alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+
+ if (board_config == ALC882_AUTO) {
/* automatic parse from the BIOS config */
- err = alc883_parse_auto_config(codec);
+ err = alc882_parse_auto_config(codec);
if (err < 0) {
alc_free(codec);
return err;
@@ -9790,7 +9869,7 @@ static int patch_alc883(struct hda_codec *codec)
printk(KERN_INFO
"hda_codec: Cannot set up configuration "
"from BIOS. Using base mode...\n");
- board_config = ALC883_3ST_2ch_DIG;
+ board_config = ALC882_3ST_DIG;
}
}
@@ -9800,63 +9879,61 @@ static int patch_alc883(struct hda_codec *codec)
return err;
}
- if (board_config != ALC883_AUTO)
- setup_preset(spec, &alc883_presets[board_config]);
+ if (board_config != ALC882_AUTO)
+ setup_preset(codec, &alc882_presets[board_config]);
- switch (codec->vendor_id) {
- case 0x10ec0888:
- if (!spec->num_adc_nids) {
- spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
- spec->adc_nids = alc883_adc_nids;
- }
- if (!spec->capsrc_nids)
- spec->capsrc_nids = alc883_capsrc_nids;
+ spec->stream_analog_playback = &alc882_pcm_analog_playback;
+ spec->stream_analog_capture = &alc882_pcm_analog_capture;
+ /* FIXME: setup DAC5 */
+ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
+ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
+
+ spec->stream_digital_playback = &alc882_pcm_digital_playback;
+ spec->stream_digital_capture = &alc882_pcm_digital_capture;
+
+ if (codec->vendor_id == 0x10ec0888)
spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */
- break;
- case 0x10ec0889:
- if (!spec->num_adc_nids) {
- spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids);
- spec->adc_nids = alc889_adc_nids;
- }
- if (!spec->capsrc_nids)
- spec->capsrc_nids = alc889_capsrc_nids;
- break;
- default:
- if (!spec->num_adc_nids) {
- spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
- spec->adc_nids = alc883_adc_nids;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ int i;
+ spec->num_adc_nids = 0;
+ for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) {
+ hda_nid_t cap;
+ hda_nid_t nid = alc882_adc_nids[i];
+ unsigned int wcap = get_wcaps(codec, nid);
+ /* get type */
+ wcap = get_wcaps_type(wcap);
+ if (wcap != AC_WID_AUD_IN)
+ continue;
+ spec->private_adc_nids[spec->num_adc_nids] = nid;
+ err = snd_hda_get_connections(codec, nid, &cap, 1);
+ if (err < 0)
+ continue;
+ spec->private_capsrc_nids[spec->num_adc_nids] = cap;
+ spec->num_adc_nids++;
}
- if (!spec->capsrc_nids)
- spec->capsrc_nids = alc883_capsrc_nids;
- break;
+ spec->adc_nids = spec->private_adc_nids;
+ spec->capsrc_nids = spec->private_capsrc_nids;
}
- spec->stream_analog_playback = &alc883_pcm_analog_playback;
- spec->stream_analog_capture = &alc883_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture;
-
- spec->stream_digital_playback = &alc883_pcm_digital_playback;
- spec->stream_digital_capture = &alc883_pcm_digital_capture;
-
- if (!spec->cap_mixer)
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x0c;
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC883_AUTO)
- spec->init_hook = alc883_auto_init;
-
+ if (board_config == ALC882_AUTO)
+ spec->init_hook = alc882_auto_init;
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (!spec->loopback.amplist)
- spec->loopback.amplist = alc883_loopbacks;
+ spec->loopback.amplist = alc882_loopbacks;
#endif
codec->proc_widget_hook = print_realtek_coef;
return 0;
}
+
/*
* ALC262 support
*/
@@ -10032,13 +10109,12 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
};
/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hp_t5735_init_hook(struct hda_codec *codec)
+static void alc262_hp_t5735_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
- alc_automute_amp(codec);
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -10195,22 +10271,20 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res)
alc262_hippo_automute(codec);
}
-static void alc262_hippo_init_hook(struct hda_codec *codec)
+static void alc262_hippo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc262_hippo_automute(codec);
}
-static void alc262_hippo1_init_hook(struct hda_codec *codec)
+static void alc262_hippo1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
- alc262_hippo_automute(codec);
}
@@ -10267,13 +10341,12 @@ static struct hda_verb alc262_tyan_verbs[] = {
};
/* unsolicited event for HP jack sensing */
-static void alc262_tyan_init_hook(struct hda_codec *codec)
+static void alc262_tyan_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x15;
- alc_automute_amp(codec);
}
@@ -10365,12 +10438,6 @@ static struct hda_verb alc262_eapd_verbs[] = {
{ }
};
-static struct hda_verb alc262_hippo_unsol_verbs[] = {
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {}
-};
-
static struct hda_verb alc262_hippo1_unsol_verbs[] = {
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -10391,14 +10458,6 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
{}
};
-static struct hda_input_mux alc262_dmic_capture_source = {
- .num_items = 2,
- .items = {
- { "Int DMic", 0x9 },
- { "Mic", 0x0 },
- },
-};
-
static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -10420,35 +10479,17 @@ static struct hda_verb alc262_toshiba_s06_verbs[] = {
{}
};
-static void alc262_dmic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x22, 0,
- AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc262_dmic_automute(codec);
- else
- alc_sku_unsol_event(codec, res);
-}
-
-static void alc262_toshiba_s06_init_hook(struct hda_codec *codec)
+static void alc262_toshiba_s06_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_pin(codec);
- alc262_dmic_automute(codec);
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x12;
+ spec->int_mic.mux_idx = 9;
+ spec->auto_mic = 1;
}
/*
@@ -10866,104 +10907,111 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
{ } /* end */
};
+/* We use two mixers depending on the output pin; 0x16 is a mono output
+ * and thus it's bound with a different mixer.
+ * This function returns which mixer amp should be used.
+ */
+static int alc262_check_volbit(hda_nid_t nid)
+{
+ if (!nid)
+ return 0;
+ else if (nid == 0x16)
+ return 2;
+ else
+ return 1;
+}
+
+static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
+ const char *pfx, int *vbits)
+{
+ char name[32];
+ unsigned long val;
+ int vbit;
+
+ vbit = alc262_check_volbit(nid);
+ if (!vbit)
+ return 0;
+ if (*vbits & vbit) /* a volume control for this mixer already there */
+ return 0;
+ *vbits |= vbit;
+ snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+ if (vbit == 2)
+ val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT);
+ else
+ val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT);
+ return add_control(spec, ALC_CTL_WIDGET_VOL, name, val);
+}
+
+static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid,
+ const char *pfx)
+{
+ char name[32];
+ unsigned long val;
+
+ if (!nid)
+ return 0;
+ snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+ if (nid == 0x16)
+ val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
+ else
+ val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val);
+}
+
/* add playback controls from the parsed DAC table */
static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
- hda_nid_t nid;
+ const char *pfx;
+ int vbits;
int err;
spec->multiout.num_dacs = 1; /* only use one dac */
spec->multiout.dac_nids = spec->private_dac_nids;
spec->multiout.dac_nids[0] = 2;
- nid = cfg->line_out_pins[0];
- if (nid) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
-
- nid = cfg->speaker_pins[0];
- if (nid) {
- if (nid == 0x16) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Speaker Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Speaker Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Speaker Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- nid = cfg->hp_pins[0];
- if (nid) {
- /* spec->multiout.hp_nid = 2; */
- if (nid == 0x16) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-static int alc262_auto_create_analog_input_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- int err;
+ if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
+ pfx = "Master";
+ else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ else
+ pfx = "Front";
+ err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[0], pfx);
+ if (err < 0)
+ return err;
+ err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[0], "Speaker");
+ if (err < 0)
+ return err;
+ err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[0], "Headphone");
+ if (err < 0)
+ return err;
- err = alc880_auto_create_analog_input_ctls(spec, cfg);
+ vbits = alc262_check_volbit(cfg->line_out_pins[0]) |
+ alc262_check_volbit(cfg->speaker_pins[0]) |
+ alc262_check_volbit(cfg->hp_pins[0]);
+ if (vbits == 1 || vbits == 2)
+ pfx = "Master"; /* only one mixer is used */
+ else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ else
+ pfx = "Front";
+ vbits = 0;
+ err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[0], pfx, &vbits);
+ if (err < 0)
+ return err;
+ err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[0], "Speaker",
+ &vbits);
+ if (err < 0)
+ return err;
+ err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[0], "Headphone",
+ &vbits);
if (err < 0)
return err;
- /* digital-mic input pin is excluded in alc880_auto_create..()
- * because it's under 0x18
- */
- if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
- cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
- struct hda_input_mux *imux = &spec->private_imux[0];
- imux->items[imux->num_items].label = "Int Mic";
- imux->items[imux->num_items].index = 0x09;
- imux->num_items++;
- }
return 0;
}
+#define alc262_auto_create_input_ctls \
+ alc880_auto_create_input_ctls
/*
* generic initialization of ADC, input mixers and output mixers
@@ -11281,7 +11329,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
- err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc262_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -11412,7 +11460,7 @@ static struct alc_config_preset alc262_presets[] = {
},
[ALC262_HIPPO] = {
.mixers = { alc262_hippo_mixer },
- .init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs},
+ .init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
@@ -11421,7 +11469,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_init_hook,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_HIPPO_1] = {
.mixers = { alc262_hippo1_mixer },
@@ -11434,7 +11483,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo1_init_hook,
+ .setup = alc262_hippo1_setup,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
@@ -11497,7 +11547,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc262_hp_t5735_init_hook,
+ .setup = alc262_hp_t5735_setup,
+ .init_hook = alc_automute_amp,
},
[ALC262_HP_RP5700] = {
.mixers = { alc262_hp_rp5700_mixer },
@@ -11528,11 +11579,13 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_init_hook,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_BENQ_T31] = {
.mixers = { alc262_benq_t31_mixer },
- .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs },
+ .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
+ alc_hp15_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
@@ -11540,7 +11593,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_init_hook,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_ULTRA] = {
.mixers = { alc262_ultra_mixer },
@@ -11592,9 +11646,9 @@ static struct alc_config_preset alc262_presets[] = {
.dig_out_nid = ALC262_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
- .input_mux = &alc262_dmic_capture_source,
- .unsol_event = alc262_toshiba_s06_unsol_event,
- .init_hook = alc262_toshiba_s06_init_hook,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc262_toshiba_s06_setup,
+ .init_hook = alc_inithook,
},
[ALC262_TOSHIBA_RX1] = {
.mixers = { alc262_toshiba_rx1_mixer },
@@ -11606,7 +11660,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_init_hook,
+ .setup = alc262_hippo_setup,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_TYAN] = {
.mixers = { alc262_tyan_mixer },
@@ -11619,7 +11674,8 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc262_tyan_init_hook,
+ .setup = alc262_tyan_setup,
+ .init_hook = alc_automute_amp,
},
};
@@ -11654,8 +11710,8 @@ static int patch_alc262(struct hda_codec *codec)
alc262_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC262_AUTO;
}
@@ -11682,7 +11738,7 @@ static int patch_alc262(struct hda_codec *codec)
}
if (board_config != ALC262_AUTO)
- setup_preset(spec, &alc262_presets[board_config]);
+ setup_preset(codec, &alc262_presets[board_config]);
spec->stream_analog_playback = &alc262_pcm_analog_playback;
spec->stream_analog_capture = &alc262_pcm_analog_capture;
@@ -11708,7 +11764,7 @@ static int patch_alc262(struct hda_codec *codec)
unsigned int wcap = get_wcaps(codec, 0x07);
/* get type */
- wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ wcap = get_wcaps_type(wcap);
if (wcap != AC_WID_AUD_IN) {
spec->adc_nids = alc262_adc_nids_alt;
spec->num_adc_nids =
@@ -11723,7 +11779,7 @@ static int patch_alc262(struct hda_codec *codec)
}
}
if (!spec->cap_mixer && !spec->no_analog)
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
if (!spec->no_analog)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -11815,14 +11871,6 @@ static struct hda_verb alc268_toshiba_verbs[] = {
{ } /* end */
};
-static struct hda_input_mux alc268_acer_lc_capture_source = {
- .num_items = 2,
- .items = {
- { "i-Mic", 0x6 },
- { "E-Mic", 0x0 },
- },
-};
-
/* Acer specific */
/* bind volumes of both NID 0x02 and 0x03 */
static struct hda_bind_ctls alc268_acer_bind_master_vol = {
@@ -11941,7 +11989,8 @@ static struct hda_verb alc268_acer_verbs[] = {
/* unsolicited event for HP jack sensing */
#define alc268_toshiba_unsol_event alc262_hippo_unsol_event
-#define alc268_toshiba_init_hook alc262_hippo_init_hook
+#define alc268_toshiba_setup alc262_hippo_setup
+#define alc268_toshiba_automute alc262_hippo_automute
static void alc268_acer_unsol_event(struct hda_codec *codec,
unsigned int res)
@@ -11971,30 +12020,33 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
AMP_IN_MUTE(0), bits);
}
-
-static void alc268_acer_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0x0 : 0x6);
-}
-
static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
- if ((res >> 26) == ALC880_HP_EVENT)
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
alc268_aspire_one_speaker_automute(codec);
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc268_acer_mic_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc268_acer_lc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x12;
+ spec->int_mic.mux_idx = 6;
+ spec->auto_mic = 1;
}
static void alc268_acer_lc_init_hook(struct hda_codec *codec)
{
alc268_aspire_one_speaker_automute(codec);
- alc268_acer_mic_automute(codec);
+ alc_mic_automute(codec);
}
static struct snd_kcontrol_new alc268_dell_mixer[] = {
@@ -12012,17 +12064,22 @@ static struct hda_verb alc268_dell_verbs[] = {
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
{ }
};
/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_dell_init_hook(struct hda_codec *codec)
+static void alc268_dell_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_pin(codec);
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
}
static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
@@ -12043,38 +12100,16 @@ static struct hda_verb alc267_quanta_il1_verbs[] = {
{ }
};
-static void alc267_quanta_il1_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL,
- present ? 0x00 : 0x01);
-}
-
-static void alc267_quanta_il1_init_hook(struct hda_codec *codec)
+static void alc267_quanta_il1_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
-
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_pin(codec);
- alc267_quanta_il1_mic_automute(codec);
-}
-
-static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_MIC_EVENT:
- alc267_quanta_il1_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
}
/*
@@ -12154,21 +12189,16 @@ static struct hda_verb alc268_volume_init_verbs[] = {
{ }
};
+static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc_mux_enum_put,
- },
+ _DEFINE_CAPSRC(1),
{ } /* end */
};
@@ -12177,18 +12207,7 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = {
HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc_mux_enum_put,
- },
+ _DEFINE_CAPSRC(2),
{ } /* end */
};
@@ -12275,26 +12294,38 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx)
{
char name[32];
+ hda_nid_t dac;
int err;
sprintf(name, "%s Playback Volume", ctlname);
- if (nid == 0x14) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(0x02, 3, idx,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else if (nid == 0x15) {
+ switch (nid) {
+ case 0x14:
+ case 0x16:
+ dac = 0x02;
+ break;
+ case 0x15:
+ dac = 0x03;
+ break;
+ default:
+ return 0;
+ }
+ if (spec->multiout.dac_nids[0] != dac &&
+ spec->multiout.dac_nids[1] != dac) {
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(0x03, 3, idx,
+ HDA_COMPOSE_AMP_VAL(dac, 3, idx,
HDA_OUTPUT));
if (err < 0)
return err;
- } else
- return -1;
+ spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+ }
+
sprintf(name, "%s Playback Switch", ctlname);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ if (nid != 0x16)
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
+ else /* mono */
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT));
if (err < 0)
return err;
return 0;
@@ -12307,14 +12338,19 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
hda_nid_t nid;
int err;
- spec->multiout.num_dacs = 2; /* only use one dac */
spec->multiout.dac_nids = spec->private_dac_nids;
- spec->multiout.dac_nids[0] = 2;
- spec->multiout.dac_nids[1] = 3;
nid = cfg->line_out_pins[0];
- if (nid)
- alc268_new_analog_output(spec, nid, "Front", 0);
+ if (nid) {
+ const char *name;
+ if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ name = "Speaker";
+ else
+ name = "Front";
+ err = alc268_new_analog_output(spec, nid, name, 0);
+ if (err < 0)
+ return err;
+ }
nid = cfg->speaker_pins[0];
if (nid == 0x1d) {
@@ -12323,16 +12359,23 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
if (err < 0)
return err;
+ } else {
+ err = alc268_new_analog_output(spec, nid, "Speaker", 0);
+ if (err < 0)
+ return err;
}
nid = cfg->hp_pins[0];
- if (nid)
- alc268_new_analog_output(spec, nid, "Headphone", 0);
+ if (nid) {
+ err = alc268_new_analog_output(spec, nid, "Headphone", 0);
+ if (err < 0)
+ return err;
+ }
nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
if (nid == 0x16) {
err = add_control(spec, ALC_CTL_WIDGET_MUTE,
"Mono Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT));
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT));
if (err < 0)
return err;
}
@@ -12340,38 +12383,46 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
}
/* create playback/capture controls for input pins */
-static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc268_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- struct hda_input_mux *imux = &spec->private_imux[0];
- int i, idx1;
+ return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24);
+}
- for (i = 0; i < AUTO_PIN_LAST; i++) {
- switch(cfg->input_pins[i]) {
- case 0x18:
- idx1 = 0; /* Mic 1 */
- break;
- case 0x19:
- idx1 = 1; /* Mic 2 */
- break;
- case 0x1a:
- idx1 = 2; /* Line In */
- break;
- case 0x1c:
- idx1 = 3; /* CD */
- break;
- case 0x12:
- case 0x13:
- idx1 = 6; /* digital mics */
- break;
- default:
- continue;
- }
- imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index = idx1;
- imux->num_items++;
+static void alc268_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type)
+{
+ int idx;
+
+ alc_set_pin_output(codec, nid, pin_type);
+ if (nid == 0x14 || nid == 0x16)
+ idx = 0;
+ else
+ idx = 1;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
+}
+
+static void alc268_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid = spec->autocfg.line_out_pins[0];
+ if (nid) {
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ alc268_auto_set_output_and_unmute(codec, nid, pin_type);
}
- return 0;
+}
+
+static void alc268_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin)
+ alc268_auto_set_output_and_unmute(codec, pin, PIN_HP);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT);
}
static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
@@ -12382,9 +12433,10 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
unsigned int dac_vol1, dac_vol2;
- if (speaker_nid) {
+ if (line_nid == 0x1d || speaker_nid == 0x1d) {
snd_hda_codec_write(codec, speaker_nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ /* mute mixer inputs from 0x1d */
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(1));
@@ -12392,6 +12444,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(1));
} else {
+ /* unmute mixer inputs from 0x1d */
snd_hda_codec_write(codec, 0x0f, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
snd_hda_codec_write(codec, 0x10, 0,
@@ -12448,7 +12501,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
- err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc268_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -12467,7 +12520,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
add_mixer(spec, alc268_beep_mixer);
add_verb(spec, alc268_volume_init_verbs);
- spec->num_mux_defs = 1;
+ spec->num_mux_defs = 2;
spec->input_mux = &spec->private_imux[0];
err = alc_auto_add_mic_boost(codec);
@@ -12479,8 +12532,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
return 1;
}
-#define alc268_auto_init_multi_out alc882_auto_init_multi_out
-#define alc268_auto_init_hp_out alc882_auto_init_hp_out
#define alc268_auto_init_analog_input alc882_auto_init_analog_input
/* init callback for auto-configuration model -- overriding the default init */
@@ -12523,8 +12574,11 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+ /* almost compatible with toshiba but with optional digital outs;
+ * auto-probing seems working fine
+ */
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
- ALC268_TOSHIBA),
+ ALC268_AUTO),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
@@ -12545,7 +12599,8 @@ static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
static struct alc_config_preset alc268_presets[] = {
[ALC267_QUANTA_IL1] = {
- .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+ .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
+ alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc267_quanta_il1_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -12555,9 +12610,9 @@ static struct alc_config_preset alc268_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
- .input_mux = &alc268_capture_source,
- .unsol_event = alc267_quanta_il1_unsol_event,
- .init_hook = alc267_quanta_il1_init_hook,
+ .unsol_event = alc_sku_unsol_event,
+ .setup = alc267_quanta_il1_setup,
+ .init_hook = alc_inithook,
},
[ALC268_3ST] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
@@ -12589,10 +12644,11 @@ static struct alc_config_preset alc268_presets[] = {
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc268_toshiba_unsol_event,
- .init_hook = alc268_toshiba_init_hook,
+ .setup = alc268_toshiba_setup,
+ .init_hook = alc268_toshiba_automute,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+ .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
@@ -12628,7 +12684,7 @@ static struct alc_config_preset alc268_presets[] = {
[ALC268_ACER_ASPIRE_ONE] = {
.mixers = { alc268_acer_aspire_one_mixer,
alc268_beep_mixer,
- alc268_capture_alt_mixer },
+ alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_aspire_one_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -12639,22 +12695,26 @@ static struct alc_config_preset alc268_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
- .input_mux = &alc268_acer_lc_capture_source,
.unsol_event = alc268_acer_lc_unsol_event,
+ .setup = alc268_acer_lc_setup,
.init_hook = alc268_acer_lc_init_hook,
},
[ALC268_DELL] = {
- .mixers = { alc268_dell_mixer, alc268_beep_mixer },
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer,
+ alc268_capture_nosrc_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_dell_verbs },
.num_dacs = ARRAY_SIZE(alc268_dac_nids),
.dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.unsol_event = alc_sku_unsol_event,
- .init_hook = alc268_dell_init_hook,
- .input_mux = &alc268_capture_source,
+ .setup = alc268_dell_setup,
+ .init_hook = alc_inithook,
},
[ALC268_ZEPTO] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
@@ -12671,8 +12731,8 @@ static struct alc_config_preset alc268_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
- .unsol_event = alc268_toshiba_unsol_event,
- .init_hook = alc268_toshiba_init_hook
+ .setup = alc268_toshiba_setup,
+ .init_hook = alc268_toshiba_automute,
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
@@ -12714,8 +12774,8 @@ static int patch_alc268(struct hda_codec *codec)
ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl);
if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC268_AUTO;
}
@@ -12734,7 +12794,7 @@ static int patch_alc268(struct hda_codec *codec)
}
if (board_config != ALC268_AUTO)
- setup_preset(spec, &alc268_presets[board_config]);
+ setup_preset(codec, &alc268_presets[board_config]);
spec->stream_analog_playback = &alc268_pcm_analog_playback;
spec->stream_analog_capture = &alc268_pcm_analog_capture;
@@ -12771,11 +12831,15 @@ static int patch_alc268(struct hda_codec *codec)
int i;
/* get type */
- wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
- if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
+ wcap = get_wcaps_type(wcap);
+ if (spec->auto_mic ||
+ wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc268_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
- add_mixer(spec, alc268_capture_alt_mixer);
+ if (spec->auto_mic || spec->input_mux->num_items == 1)
+ add_mixer(spec, alc268_capture_nosrc_mixer);
+ else
+ add_mixer(spec, alc268_capture_alt_mixer);
} else {
spec->adc_nids = alc268_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
@@ -12786,6 +12850,8 @@ static int patch_alc268(struct hda_codec *codec)
for (i = 0; i < spec->num_adc_nids; i++)
snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
0, AC_VERB_SET_CONNECT_SEL,
+ i < spec->num_mux_defs ?
+ spec->input_mux[i].items[0].index :
spec->input_mux->items[0].index);
}
@@ -12820,22 +12886,6 @@ static hda_nid_t alc269_capsrc_nids[1] = {
* not a mux!
*/
-static struct hda_input_mux alc269_eeepc_dmic_capture_source = {
- .num_items = 2,
- .items = {
- { "i-Mic", 0x5 },
- { "e-Mic", 0x0 },
- },
-};
-
-static struct hda_input_mux alc269_eeepc_amic_capture_source = {
- .num_items = 2,
- .items = {
- { "i-Mic", 0x1 },
- { "e-Mic", 0x0 },
- },
-};
-
#define alc269_modes alc260_modes
#define alc269_capture_source alc880_lg_lw_capture_source
@@ -12997,16 +13047,6 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
AC_VERB_SET_PROC_COEF, 0x480);
}
-static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
-}
-
static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
{
unsigned int present_laptop;
@@ -13033,10 +13073,14 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
- if ((res >> 26) == ALC880_HP_EVENT)
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
alc269_quanta_fl1_speaker_automute(codec);
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc269_quanta_fl1_mic_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
}
static void alc269_lifebook_unsol_event(struct hda_codec *codec,
@@ -13048,10 +13092,20 @@ static void alc269_lifebook_unsol_event(struct hda_codec *codec,
alc269_lifebook_mic_autoswitch(codec);
}
+static void alc269_quanta_fl1_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
+}
+
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
alc269_quanta_fl1_speaker_automute(codec);
- alc269_quanta_fl1_mic_automute(codec);
+ alc_mic_automute(codec);
}
static void alc269_lifebook_init_hook(struct hda_codec *codec)
@@ -13096,60 +13150,44 @@ static void alc269_speaker_automute(struct hda_codec *codec)
AMP_IN_MUTE(0), bits);
}
-static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x23, 0,
- AC_VERB_SET_CONNECT_SEL, (present ? 0 : 5));
-}
-
-static void alc269_eeepc_amic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-}
-
/* unsolicited event for HP jack sensing */
-static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
+static void alc269_eeepc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
- if ((res >> 26) == ALC880_HP_EVENT)
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
alc269_speaker_automute(codec);
-
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc269_eeepc_dmic_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc_mic_automute(codec);
+ break;
+ }
}
-static void alc269_eeepc_dmic_inithook(struct hda_codec *codec)
+static void alc269_eeepc_dmic_setup(struct hda_codec *codec)
{
- alc269_speaker_automute(codec);
- alc269_eeepc_dmic_automute(codec);
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x12;
+ spec->int_mic.mux_idx = 5;
+ spec->auto_mic = 1;
}
-/* unsolicited event for HP jack sensing */
-static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc269_eeepc_amic_setup(struct hda_codec *codec)
{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc269_speaker_automute(codec);
-
- if ((res >> 26) == ALC880_MIC_EVENT)
- alc269_eeepc_amic_automute(codec);
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
}
-static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
+static void alc269_eeepc_inithook(struct hda_codec *codec)
{
alc269_speaker_automute(codec);
- alc269_eeepc_amic_automute(codec);
+ alc_mic_automute(codec);
}
/*
@@ -13222,89 +13260,10 @@ static struct hda_verb alc269_init_verbs[] = {
{ }
};
-/* add playback controls from the parsed DAC table */
-static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg)
-{
- hda_nid_t nid;
- int err;
-
- spec->multiout.num_dacs = 1; /* only use one dac */
- spec->multiout.dac_nids = spec->private_dac_nids;
- spec->multiout.dac_nids[0] = 2;
-
- nid = cfg->line_out_pins[0];
- if (nid) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
-
- nid = cfg->speaker_pins[0];
- if (nid) {
- if (!cfg->line_out_pins[0]) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Speaker Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- if (nid == 0x16) {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Speaker Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Speaker Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- nid = cfg->hp_pins[0];
- if (nid) {
- /* spec->multiout.hp_nid = 2; */
- if (!cfg->line_out_pins[0] && !cfg->speaker_pins[0]) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Headphone Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- if (nid == 0x16) {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- } else {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- }
- return 0;
-}
-
-#define alc269_auto_create_analog_input_ctls \
- alc262_auto_create_analog_input_ctls
+#define alc269_auto_create_multi_out_ctls \
+ alc268_auto_create_multi_out_ctls
+#define alc269_auto_create_input_ctls \
+ alc268_auto_create_input_ctls
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc269_loopbacks alc880_loopbacks
@@ -13354,7 +13313,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
- err = alc269_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc269_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -13379,15 +13338,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
return err;
if (!spec->cap_mixer && !spec->no_analog)
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
alc_ssid_check(codec, 0x15, 0x1b, 0x14);
return 1;
}
-#define alc269_auto_init_multi_out alc882_auto_init_multi_out
-#define alc269_auto_init_hp_out alc882_auto_init_hp_out
+#define alc269_auto_init_multi_out alc268_auto_init_multi_out
+#define alc269_auto_init_hp_out alc268_auto_init_hp_out
#define alc269_auto_init_analog_input alc882_auto_init_analog_input
@@ -13455,6 +13414,7 @@ static struct alc_config_preset alc269_presets[] = {
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_quanta_fl1_unsol_event,
+ .setup = alc269_quanta_fl1_setup,
.init_hook = alc269_quanta_fl1_init_hook,
},
[ALC269_ASUS_EEEPC_P703] = {
@@ -13467,9 +13427,9 @@ static struct alc_config_preset alc269_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
- .input_mux = &alc269_eeepc_amic_capture_source,
- .unsol_event = alc269_eeepc_amic_unsol_event,
- .init_hook = alc269_eeepc_amic_inithook,
+ .unsol_event = alc269_eeepc_unsol_event,
+ .setup = alc269_eeepc_amic_setup,
+ .init_hook = alc269_eeepc_inithook,
},
[ALC269_ASUS_EEEPC_P901] = {
.mixers = { alc269_eeepc_mixer },
@@ -13481,9 +13441,9 @@ static struct alc_config_preset alc269_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
- .input_mux = &alc269_eeepc_dmic_capture_source,
- .unsol_event = alc269_eeepc_dmic_unsol_event,
- .init_hook = alc269_eeepc_dmic_inithook,
+ .unsol_event = alc269_eeepc_unsol_event,
+ .setup = alc269_eeepc_dmic_setup,
+ .init_hook = alc269_eeepc_inithook,
},
[ALC269_FUJITSU] = {
.mixers = { alc269_fujitsu_mixer },
@@ -13495,9 +13455,9 @@ static struct alc_config_preset alc269_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
- .input_mux = &alc269_eeepc_dmic_capture_source,
- .unsol_event = alc269_eeepc_dmic_unsol_event,
- .init_hook = alc269_eeepc_dmic_inithook,
+ .unsol_event = alc269_eeepc_unsol_event,
+ .setup = alc269_eeepc_dmic_setup,
+ .init_hook = alc269_eeepc_inithook,
},
[ALC269_LIFEBOOK] = {
.mixers = { alc269_lifebook_mixer },
@@ -13532,8 +13492,8 @@ static int patch_alc269(struct hda_codec *codec)
alc269_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC269_AUTO;
}
@@ -13558,7 +13518,7 @@ static int patch_alc269(struct hda_codec *codec)
}
if (board_config != ALC269_AUTO)
- setup_preset(spec, &alc269_presets[board_config]);
+ setup_preset(codec, &alc269_presets[board_config]);
if (codec->subsystem_id == 0x17aa3bf8) {
/* Due to a hardware problem on Lenovo Ideadpad, we need to
@@ -13577,7 +13537,7 @@ static int patch_alc269(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
spec->capsrc_nids = alc269_capsrc_nids;
if (!spec->cap_mixer)
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
spec->vmaster_nid = 0x02;
@@ -14127,23 +14087,23 @@ static struct hda_verb alc861_auto_init_verbs[] = {
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set Mic 1 */
@@ -14215,64 +14175,96 @@ static struct hda_input_mux alc861_capture_source = {
},
};
+static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t mix, srcs[5];
+ int i, j, num;
+
+ if (snd_hda_get_connections(codec, pin, &mix, 1) != 1)
+ return 0;
+ num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
+ if (num < 0)
+ return 0;
+ for (i = 0; i < num; i++) {
+ unsigned int type;
+ type = get_wcaps_type(get_wcaps(codec, srcs[i]));
+ if (type != AC_WID_AUD_OUT)
+ continue;
+ for (j = 0; j < spec->multiout.num_dacs; j++)
+ if (spec->multiout.dac_nids[j] == srcs[i])
+ break;
+ if (j >= spec->multiout.num_dacs)
+ return srcs[i];
+ }
+ return 0;
+}
+
/* fill in the dac_nids table from the parsed pin configuration */
-static int alc861_auto_fill_dac_nids(struct alc_spec *spec,
+static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
+ struct alc_spec *spec = codec->spec;
int i;
- hda_nid_t nid;
+ hda_nid_t nid, dac;
spec->multiout.dac_nids = spec->private_dac_nids;
for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
- if (nid) {
- if (i >= ARRAY_SIZE(alc861_dac_nids))
- continue;
- spec->multiout.dac_nids[i] = alc861_dac_nids[i];
- }
+ dac = alc861_look_for_dac(codec, nid);
+ if (!dac)
+ continue;
+ spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
}
- spec->multiout.num_dacs = cfg->line_outs;
return 0;
}
+static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx,
+ hda_nid_t nid, unsigned int chs)
+{
+ char name[32];
+ snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+ return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
/* add playback controls from the parsed DAC table */
-static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
+ struct alc_spec *spec = codec->spec;
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
hda_nid_t nid;
- int i, idx, err;
+ int i, err;
+
+ if (cfg->line_outs == 1) {
+ const char *pfx = NULL;
+ if (!cfg->hp_outs)
+ pfx = "Master";
+ else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ if (pfx) {
+ nid = spec->multiout.dac_nids[0];
+ return alc861_create_out_sw(codec, pfx, nid, 3);
+ }
+ }
for (i = 0; i < cfg->line_outs; i++) {
nid = spec->multiout.dac_nids[i];
if (!nid)
continue;
- if (nid == 0x05) {
+ if (i == 2) {
/* Center/LFE */
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0,
- HDA_OUTPUT));
+ err = alc861_create_out_sw(codec, "Center", nid, 1);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
+ err = alc861_create_out_sw(codec, "LFE", nid, 2);
if (err < 0)
return err;
} else {
- for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1;
- idx++)
- if (nid == alc861_dac_nids[idx])
- break;
- sprintf(name, "%s Playback Switch", chname[idx]);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
+ err = alc861_create_out_sw(codec, chname[i], nid, 3);
if (err < 0)
return err;
}
@@ -14280,8 +14272,9 @@ static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
return 0;
}
-static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
+static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
{
+ struct alc_spec *spec = codec->spec;
int err;
hda_nid_t nid;
@@ -14289,70 +14282,49 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
return 0;
if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
- nid = 0x03;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Headphone Playback Switch",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- spec->multiout.hp_nid = nid;
+ nid = alc861_look_for_dac(codec, pin);
+ if (nid) {
+ err = alc861_create_out_sw(codec, "Headphone", nid, 3);
+ if (err < 0)
+ return err;
+ spec->multiout.hp_nid = nid;
+ }
}
return 0;
}
/* create playback/capture controls for input pins */
-static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc861_auto_create_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- struct hda_input_mux *imux = &spec->private_imux[0];
- int i, err, idx, idx1;
-
- for (i = 0; i < AUTO_PIN_LAST; i++) {
- switch (cfg->input_pins[i]) {
- case 0x0c:
- idx1 = 1;
- idx = 2; /* Line In */
- break;
- case 0x0f:
- idx1 = 2;
- idx = 2; /* Line In */
- break;
- case 0x0d:
- idx1 = 0;
- idx = 1; /* Mic In */
- break;
- case 0x10:
- idx1 = 3;
- idx = 1; /* Mic In */
- break;
- case 0x11:
- idx1 = 4;
- idx = 0; /* CD */
- break;
- default:
- continue;
- }
-
- err = new_analog_input(spec, cfg->input_pins[i],
- auto_pin_cfg_labels[i], idx, 0x15);
- if (err < 0)
- return err;
-
- imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index = idx1;
- imux->num_items++;
- }
- return 0;
+ return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x08, 0);
}
static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid,
- int pin_type, int dac_idx)
+ int pin_type, hda_nid_t dac)
{
+ hda_nid_t mix, srcs[5];
+ int i, num;
+
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_type);
- snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
+ if (snd_hda_get_connections(codec, nid, &mix, 1) != 1)
+ return;
+ num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
+ if (num < 0)
+ return;
+ for (i = 0; i < num; i++) {
+ unsigned int mute;
+ if (srcs[i] == dac || srcs[i] == 0x15)
+ mute = AMP_IN_UNMUTE(i);
+ else
+ mute = AMP_IN_MUTE(i);
+ snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ mute);
+ }
}
static void alc861_auto_init_multi_out(struct hda_codec *codec)
@@ -14375,12 +14347,13 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec)
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
+ if (pin)
alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.dac_nids[0]);
+ spec->multiout.hp_nid);
pin = spec->autocfg.speaker_pins[0];
if (pin)
- alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ spec->multiout.dac_nids[0]);
}
static void alc861_auto_init_analog_input(struct hda_codec *codec)
@@ -14412,16 +14385,16 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
- err = alc861_auto_fill_dac_nids(spec, &spec->autocfg);
+ err = alc861_auto_fill_dac_nids(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
if (err < 0)
return err;
- err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc861_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -14440,7 +14413,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
spec->adc_nids = alc861_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
alc_ssid_check(codec, 0x0e, 0x0f, 0x0b);
@@ -14633,8 +14606,8 @@ static int patch_alc861(struct hda_codec *codec)
alc861_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC861_AUTO;
}
@@ -14659,7 +14632,7 @@ static int patch_alc861(struct hda_codec *codec)
}
if (board_config != ALC861_AUTO)
- setup_preset(spec, &alc861_presets[board_config]);
+ setup_preset(codec, &alc861_presets[board_config]);
spec->stream_analog_playback = &alc861_pcm_analog_playback;
spec->stream_analog_capture = &alc861_pcm_analog_capture;
@@ -15062,12 +15035,15 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
HDA_AMP_MUTE, bits);
}
-static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
+static void alc861vd_lenovo_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
-
spec->autocfg.hp_pins[0] = 0x1b;
spec->autocfg.speaker_pins[0] = 0x14;
+}
+
+static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
+{
alc_automute_amp(codec);
alc861vd_lenovo_mic_automute(codec);
}
@@ -15131,13 +15107,12 @@ static struct hda_verb alc861vd_dallas_verbs[] = {
};
/* toggle speaker-output according to the hp-jack state */
-static void alc861vd_dallas_init_hook(struct hda_codec *codec)
+static void alc861vd_dallas_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
- alc_automute_amp(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -15251,6 +15226,7 @@ static struct alc_config_preset alc861vd_presets[] = {
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
+ .setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
[ALC861VD_DALLAS] = {
@@ -15262,7 +15238,8 @@ static struct alc_config_preset alc861vd_presets[] = {
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc861vd_dallas_init_hook,
+ .setup = alc861vd_dallas_setup,
+ .init_hook = alc_automute_amp,
},
[ALC861VD_HP] = {
.mixers = { alc861vd_hp_mixer },
@@ -15274,7 +15251,8 @@ static struct alc_config_preset alc861vd_presets[] = {
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
.unsol_event = alc_automute_amp_unsol_event,
- .init_hook = alc861vd_dallas_init_hook,
+ .setup = alc861vd_dallas_setup,
+ .init_hook = alc_automute_amp,
},
[ALC660VD_ASUS_V1S] = {
.mixers = { alc861vd_lenovo_mixer },
@@ -15289,6 +15267,7 @@ static struct alc_config_preset alc861vd_presets[] = {
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
+ .setup = alc861vd_lenovo_setup,
.init_hook = alc861vd_lenovo_init_hook,
},
};
@@ -15296,6 +15275,13 @@ static struct alc_config_preset alc861vd_presets[] = {
/*
* BIOS auto configuration
*/
+static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+}
+
+
static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type, int dac_idx)
{
@@ -15330,7 +15316,6 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
}
-#define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid)
#define ALC861VD_PIN_CD_NID ALC880_PIN_CD_NID
static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
@@ -15340,7 +15325,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
- if (alc861vd_is_input_pin(nid)) {
+ if (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC861VD_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -15404,13 +15389,25 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
if (err < 0)
return err;
} else {
- sprintf(name, "%s Playback Volume", chname[i]);
+ const char *pfx;
+ if (cfg->line_outs == 1 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ if (!cfg->hp_pins)
+ pfx = "Speaker";
+ else
+ pfx = "PCM";
+ } else
+ pfx = chname[i];
+ sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", chname[i]);
+ if (cfg->line_outs == 1 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_BIND_MUTE, name,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
HDA_INPUT));
@@ -15503,7 +15500,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
"Headphone");
if (err < 0)
return err;
- err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ err = alc861vd_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -15557,8 +15554,8 @@ static int patch_alc861vd(struct hda_codec *codec)
alc861vd_cfg_tbl);
if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC861VD_AUTO;
}
@@ -15583,7 +15580,7 @@ static int patch_alc861vd(struct hda_codec *codec)
}
if (board_config != ALC861VD_AUTO)
- setup_preset(spec, &alc861vd_presets[board_config]);
+ setup_preset(codec, &alc861vd_presets[board_config]);
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
@@ -15603,7 +15600,7 @@ static int patch_alc861vd(struct hda_codec *codec)
if (!spec->capsrc_nids)
spec->capsrc_nids = alc861vd_capsrc_nids;
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
spec->vmaster_nid = 0x02;
@@ -15644,9 +15641,9 @@ static hda_nid_t alc272_dac_nids[2] = {
0x02, 0x03
};
-static hda_nid_t alc662_adc_nids[1] = {
+static hda_nid_t alc662_adc_nids[2] = {
/* ADC1-2 */
- 0x09,
+ 0x09, 0x08
};
static hda_nid_t alc272_adc_nids[1] = {
@@ -15654,7 +15651,7 @@ static hda_nid_t alc272_adc_nids[1] = {
0x08,
};
-static hda_nid_t alc662_capsrc_nids[1] = { 0x22 };
+static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
static hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
@@ -15678,14 +15675,6 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = {
},
};
-static struct hda_input_mux alc662_eeepc_capture_source = {
- .num_items = 2,
- .items = {
- { "i-Mic", 0x1 },
- { "e-Mic", 0x0 },
- },
-};
-
static struct hda_input_mux alc663_capture_source = {
.num_items = 3,
.items = {
@@ -15695,23 +15684,7 @@ static struct hda_input_mux alc663_capture_source = {
},
};
-static struct hda_input_mux alc663_m51va_capture_source = {
- .num_items = 2,
- .items = {
- { "Ext-Mic", 0x0 },
- { "D-Mic", 0x9 },
- },
-};
-
-#if 1 /* set to 0 for testing other input sources below */
-static struct hda_input_mux alc272_nc10_capture_source = {
- .num_items = 2,
- .items = {
- { "Autoselect Mic", 0x0 },
- { "Internal Mic", 0x1 },
- },
-};
-#else
+#if 0 /* set to 1 for testing other input sources below */
static struct hda_input_mux alc272_nc10_capture_source = {
.num_items = 16,
.items = {
@@ -16380,47 +16353,44 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
alc662_lenovo_101e_ispeaker_automute(codec);
}
-static void alc662_eeepc_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
- snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-}
-
/* unsolicited event for HP jack sensing */
static void alc662_eeepc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_MIC_EVENT)
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
else
alc262_hippo_unsol_event(codec, res);
}
+static void alc662_eeepc_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ alc262_hippo1_setup(codec);
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x19;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
+}
+
static void alc662_eeepc_inithook(struct hda_codec *codec)
{
- alc262_hippo1_init_hook(codec);
- alc662_eeepc_mic_automute(codec);
+ alc262_hippo_automute(codec);
+ alc_mic_automute(codec);
}
-static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
+static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
spec->autocfg.hp_pins[0] = 0x14;
spec->autocfg.speaker_pins[0] = 0x1b;
- alc262_hippo_master_update(codec);
}
+#define alc662_eeepc_ep20_inithook alc262_hippo_master_update
+
static void alc663_m51va_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -16531,23 +16501,6 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
}
}
-static void alc663_m51va_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
- snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
- snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- 0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
-}
-
static void alc663_m51va_unsol_event(struct hda_codec *codec,
unsigned int res)
{
@@ -16556,36 +16509,32 @@ static void alc663_m51va_unsol_event(struct hda_codec *codec,
alc663_m51va_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc663_m51va_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+static void alc663_m51va_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->ext_mic.pin = 0x18;
+ spec->ext_mic.mux_idx = 0;
+ spec->int_mic.pin = 0x12;
+ spec->int_mic.mux_idx = 1;
+ spec->auto_mic = 1;
+}
+
static void alc663_m51va_inithook(struct hda_codec *codec)
{
alc663_m51va_speaker_automute(codec);
- alc663_m51va_mic_automute(codec);
+ alc_mic_automute(codec);
}
/* ***************** Mode1 ******************************/
-static void alc663_mode1_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc663_m51va_speaker_automute(codec);
- break;
- case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
- break;
- }
-}
+#define alc663_mode1_unsol_event alc663_m51va_unsol_event
+#define alc663_mode1_setup alc663_m51va_setup
+#define alc663_mode1_inithook alc663_m51va_inithook
-static void alc663_mode1_inithook(struct hda_codec *codec)
-{
- alc663_m51va_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
-}
/* ***************** Mode2 ******************************/
static void alc662_mode2_unsol_event(struct hda_codec *codec,
unsigned int res)
@@ -16595,15 +16544,17 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec,
alc662_f5z_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc662_mode2_setup alc663_m51va_setup
+
static void alc662_mode2_inithook(struct hda_codec *codec)
{
alc662_f5z_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
/* ***************** Mode3 ******************************/
static void alc663_mode3_unsol_event(struct hda_codec *codec,
@@ -16614,15 +16565,17 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec,
alc663_two_hp_m1_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_mode3_setup alc663_m51va_setup
+
static void alc663_mode3_inithook(struct hda_codec *codec)
{
alc663_two_hp_m1_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
/* ***************** Mode4 ******************************/
static void alc663_mode4_unsol_event(struct hda_codec *codec,
@@ -16633,15 +16586,17 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec,
alc663_21jd_two_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_mode4_setup alc663_m51va_setup
+
static void alc663_mode4_inithook(struct hda_codec *codec)
{
alc663_21jd_two_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
/* ***************** Mode5 ******************************/
static void alc663_mode5_unsol_event(struct hda_codec *codec,
@@ -16652,15 +16607,17 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec,
alc663_15jd_two_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_mode5_setup alc663_m51va_setup
+
static void alc663_mode5_inithook(struct hda_codec *codec)
{
alc663_15jd_two_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
/* ***************** Mode6 ******************************/
static void alc663_mode6_unsol_event(struct hda_codec *codec,
@@ -16671,15 +16628,17 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec,
alc663_two_hp_m2_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_mode6_setup alc663_m51va_setup
+
static void alc663_mode6_inithook(struct hda_codec *codec)
{
alc663_two_hp_m2_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
static void alc663_g71v_hp_automute(struct hda_codec *codec)
@@ -16721,16 +16680,18 @@ static void alc663_g71v_unsol_event(struct hda_codec *codec,
alc663_g71v_front_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_g71v_setup alc663_m51va_setup
+
static void alc663_g71v_inithook(struct hda_codec *codec)
{
alc663_g71v_front_automute(codec);
alc663_g71v_hp_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
static void alc663_g50v_unsol_event(struct hda_codec *codec,
@@ -16741,15 +16702,17 @@ static void alc663_g50v_unsol_event(struct hda_codec *codec,
alc663_m51va_speaker_automute(codec);
break;
case ALC880_MIC_EVENT:
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
break;
}
}
+#define alc663_g50v_setup alc663_m51va_setup
+
static void alc663_g50v_inithook(struct hda_codec *codec)
{
alc663_m51va_speaker_automute(codec);
- alc662_eeepc_mic_automute(codec);
+ alc_mic_automute(codec);
}
static struct snd_kcontrol_new alc662_ecs_mixer[] = {
@@ -16953,8 +16916,8 @@ static struct alc_config_preset alc662_presets[] = {
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc662_eeepc_unsol_event,
+ .setup = alc662_eeepc_setup,
.init_hook = alc662_eeepc_inithook,
},
[ALC662_ASUS_EEEPC_EP20] = {
@@ -16968,6 +16931,7 @@ static struct alc_config_preset alc662_presets[] = {
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
.unsol_event = alc662_eeepc_unsol_event,
+ .setup = alc662_eeepc_ep20_setup,
.init_hook = alc662_eeepc_ep20_inithook,
},
[ALC662_ECS] = {
@@ -16978,8 +16942,8 @@ static struct alc_config_preset alc662_presets[] = {
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc662_eeepc_unsol_event,
+ .setup = alc662_eeepc_setup,
.init_hook = alc662_eeepc_inithook,
},
[ALC663_ASUS_M51VA] = {
@@ -16990,8 +16954,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc663_m51va_capture_source,
.unsol_event = alc663_m51va_unsol_event,
+ .setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC663_ASUS_G71V] = {
@@ -17002,8 +16966,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_g71v_unsol_event,
+ .setup = alc663_g71v_setup,
.init_hook = alc663_g71v_inithook,
},
[ALC663_ASUS_H13] = {
@@ -17013,7 +16977,6 @@ static struct alc_config_preset alc662_presets[] = {
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc663_m51va_capture_source,
.unsol_event = alc663_m51va_unsol_event,
.init_hook = alc663_m51va_inithook,
},
@@ -17027,6 +16990,7 @@ static struct alc_config_preset alc662_presets[] = {
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc663_capture_source,
.unsol_event = alc663_g50v_unsol_event,
+ .setup = alc663_g50v_setup,
.init_hook = alc663_g50v_inithook,
},
[ALC663_ASUS_MODE1] = {
@@ -17040,8 +17004,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_mode1_unsol_event,
+ .setup = alc663_mode1_setup,
.init_hook = alc663_mode1_inithook,
},
[ALC662_ASUS_MODE2] = {
@@ -17054,8 +17018,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc662_mode2_unsol_event,
+ .setup = alc662_mode2_setup,
.init_hook = alc662_mode2_inithook,
},
[ALC663_ASUS_MODE3] = {
@@ -17069,8 +17033,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_mode3_unsol_event,
+ .setup = alc663_mode3_setup,
.init_hook = alc663_mode3_inithook,
},
[ALC663_ASUS_MODE4] = {
@@ -17084,8 +17048,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_mode4_unsol_event,
+ .setup = alc663_mode4_setup,
.init_hook = alc663_mode4_inithook,
},
[ALC663_ASUS_MODE5] = {
@@ -17099,8 +17063,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_mode5_unsol_event,
+ .setup = alc663_mode5_setup,
.init_hook = alc663_mode5_inithook,
},
[ALC663_ASUS_MODE6] = {
@@ -17114,8 +17078,8 @@ static struct alc_config_preset alc662_presets[] = {
.dig_out_nid = ALC662_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc662_eeepc_capture_source,
.unsol_event = alc663_mode6_unsol_event,
+ .setup = alc663_mode6_setup,
.init_hook = alc663_mode6_inithook,
},
[ALC272_DELL] = {
@@ -17129,8 +17093,8 @@ static struct alc_config_preset alc662_presets[] = {
.num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
.capsrc_nids = alc272_capsrc_nids,
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc663_m51va_capture_source,
.unsol_event = alc663_m51va_unsol_event,
+ .setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC272_DELL_ZM1] = {
@@ -17141,11 +17105,11 @@ static struct alc_config_preset alc662_presets[] = {
.dac_nids = alc662_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.adc_nids = alc662_adc_nids,
- .num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
+ .num_adc_nids = 1,
.capsrc_nids = alc662_capsrc_nids,
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc663_m51va_capture_source,
.unsol_event = alc663_m51va_unsol_event,
+ .setup = alc663_m51va_setup,
.init_hook = alc663_m51va_inithook,
},
[ALC272_SAMSUNG_NC10] = {
@@ -17156,8 +17120,9 @@ static struct alc_config_preset alc662_presets[] = {
.dac_nids = alc272_dac_nids,
.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
.channel_mode = alc662_3ST_2ch_modes,
- .input_mux = &alc272_nc10_capture_source,
+ /*.input_mux = &alc272_nc10_capture_source,*/
.unsol_event = alc663_mode4_unsol_event,
+ .setup = alc663_mode4_setup,
.init_hook = alc663_mode4_inithook,
},
};
@@ -17209,13 +17174,25 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
if (err < 0)
return err;
} else {
- sprintf(name, "%s Playback Volume", chname[i]);
+ const char *pfx;
+ if (cfg->line_outs == 1 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ if (!cfg->hp_pins)
+ pfx = "Speaker";
+ else
+ pfx = "PCM";
+ } else
+ pfx = chname[i];
+ sprintf(name, "%s Playback Volume", pfx);
err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", chname[i]);
+ if (cfg->line_outs == 1 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+ pfx = "Speaker";
+ sprintf(name, "%s Playback Switch", pfx);
err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
3, 0, HDA_INPUT));
@@ -17277,62 +17254,9 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
return 0;
}
-/* return the index of the src widget from the connection list of the nid.
- * return -1 if not found
- */
-static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid,
- hda_nid_t src)
-{
- hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
- int i, conns;
-
- conns = snd_hda_get_connections(codec, nid, conn_list,
- ARRAY_SIZE(conn_list));
- if (conns < 0)
- return -1;
- for (i = 0; i < conns; i++)
- if (conn_list[i] == src)
- return i;
- return -1;
-}
-
-static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
-{
- unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
- return (pincap & AC_PINCAP_IN) != 0;
-}
-
/* create playback/capture controls for input pins */
-static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg)
-{
- struct alc_spec *spec = codec->spec;
- struct hda_input_mux *imux = &spec->private_imux[0];
- int i, err, idx;
-
- for (i = 0; i < AUTO_PIN_LAST; i++) {
- if (alc662_is_input_pin(codec, cfg->input_pins[i])) {
- idx = alc662_input_pin_idx(codec, 0x0b,
- cfg->input_pins[i]);
- if (idx >= 0) {
- err = new_analog_input(spec, cfg->input_pins[i],
- auto_pin_cfg_labels[i],
- idx, 0x0b);
- if (err < 0)
- return err;
- }
- idx = alc662_input_pin_idx(codec, 0x22,
- cfg->input_pins[i]);
- if (idx >= 0) {
- imux->items[imux->num_items].label =
- auto_pin_cfg_labels[i];
- imux->items[imux->num_items].index = idx;
- imux->num_items++;
- }
- }
- }
- return 0;
-}
+#define alc662_auto_create_input_ctls \
+ alc880_auto_create_input_ctls
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
@@ -17386,7 +17310,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = spec->autocfg.input_pins[i];
- if (alc662_is_input_pin(codec, nid)) {
+ if (alc_is_input_pin(codec, nid)) {
alc_set_input_pin(codec, nid, i);
if (nid != ALC662_PIN_CD_NID &&
(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -17427,7 +17351,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
"Headphone");
if (err < 0)
return err;
- err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg);
+ err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -17484,8 +17408,8 @@ static int patch_alc662(struct hda_codec *codec)
alc662_models,
alc662_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for %s, "
- "trying auto-probe from BIOS...\n", codec->chip_name);
+ printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
board_config = ALC662_AUTO;
}
@@ -17510,7 +17434,7 @@ static int patch_alc662(struct hda_codec *codec)
}
if (board_config != ALC662_AUTO)
- setup_preset(spec, &alc662_presets[board_config]);
+ setup_preset(codec, &alc662_presets[board_config]);
spec->stream_analog_playback = &alc662_pcm_analog_playback;
spec->stream_analog_capture = &alc662_pcm_analog_capture;
@@ -17526,7 +17450,7 @@ static int patch_alc662(struct hda_codec *codec)
spec->capsrc_nids = alc662_capsrc_nids;
if (!spec->cap_mixer)
- set_capture_mixer(spec);
+ set_capture_mixer(codec);
if (codec->vendor_id == 0x10ec0662)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
else
@@ -17562,23 +17486,23 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
{ .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
{ .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
- .patch = patch_alc883 },
+ .patch = patch_alc882 },
{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
.patch = patch_alc662 },
{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
- { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
+ { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
{ .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
- .patch = patch_alc882 }, /* should be patch_alc883() in future */
+ .patch = patch_alc882 },
{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
- .patch = patch_alc882 }, /* should be patch_alc883() in future */
+ .patch = patch_alc882 },
{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
- .patch = patch_alc883 },
- { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
- { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
+ .patch = patch_alc882 },
+ { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
+ { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6990cfc..e31e53d 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -40,6 +40,8 @@ enum {
STAC_INSERT_EVENT,
STAC_PWR_EVENT,
STAC_HP_EVENT,
+ STAC_LO_EVENT,
+ STAC_MIC_EVENT,
};
enum {
@@ -81,6 +83,7 @@ enum {
STAC_DELL_M6_DMIC,
STAC_DELL_M6_BOTH,
STAC_DELL_EQ,
+ STAC_ALIENWARE_M17X,
STAC_92HD73XX_MODELS
};
@@ -177,6 +180,12 @@ struct sigmatel_jack {
struct snd_jack *jack;
};
+struct sigmatel_mic_route {
+ hda_nid_t pin;
+ unsigned char mux_idx;
+ unsigned char dmux_idx;
+};
+
struct sigmatel_spec {
struct snd_kcontrol_new *mixers[4];
unsigned int num_mixers;
@@ -188,6 +197,7 @@ struct sigmatel_spec {
unsigned int hp_detect: 1;
unsigned int spdif_mute: 1;
unsigned int check_volume_offset:1;
+ unsigned int auto_mic:1;
/* gpio lines */
unsigned int eapd_mask;
@@ -219,7 +229,6 @@ struct sigmatel_spec {
/* playback */
struct hda_input_mux *mono_mux;
- struct hda_input_mux *amp_mux;
unsigned int cur_mmux;
struct hda_multi_out multiout;
hda_nid_t dac_nids[5];
@@ -239,6 +248,15 @@ struct sigmatel_spec {
unsigned int num_dmuxes;
hda_nid_t *smux_nids;
unsigned int num_smuxes;
+ unsigned int num_analog_muxes;
+
+ unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */
+ unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
+ unsigned int num_caps; /* number of capture volume/switch elements */
+
+ struct sigmatel_mic_route ext_mic;
+ struct sigmatel_mic_route int_mic;
+
const char **spdif_labels;
hda_nid_t dig_in_nid;
@@ -263,7 +281,6 @@ struct sigmatel_spec {
unsigned int cur_smux[2];
unsigned int cur_amux;
hda_nid_t *amp_nids;
- unsigned int num_amps;
unsigned int powerdown_adcs;
/* i/o switches */
@@ -282,7 +299,6 @@ struct sigmatel_spec {
struct hda_input_mux private_dimux;
struct hda_input_mux private_imux;
struct hda_input_mux private_smux;
- struct hda_input_mux private_amp_mux;
struct hda_input_mux private_mono_mux;
};
@@ -311,11 +327,6 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = {
0x1a, 0x1b
};
-#define DELL_M6_AMP 2
-static hda_nid_t stac92hd73xx_amp_nids[3] = {
- 0x0b, 0x0c, 0x0e
-};
-
#define STAC92HD73XX_NUM_DMICS 2
static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
0x13, 0x14, 0
@@ -323,8 +334,8 @@ static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
#define STAC92HD73_DAC_COUNT 5
-static hda_nid_t stac92hd73xx_mux_nids[4] = {
- 0x28, 0x29, 0x2a, 0x2b,
+static hda_nid_t stac92hd73xx_mux_nids[2] = {
+ 0x20, 0x21,
};
static hda_nid_t stac92hd73xx_dmux_nids[2] = {
@@ -335,14 +346,16 @@ static hda_nid_t stac92hd73xx_smux_nids[2] = {
0x22, 0x23,
};
-#define STAC92HD83XXX_NUM_DMICS 2
-static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
- 0x11, 0x12, 0
+#define STAC92HD73XX_NUM_CAPS 2
+static unsigned long stac92hd73xx_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
};
+#define stac92hd73xx_capsws stac92hd73xx_capvols
#define STAC92HD83_DAC_COUNT 3
-static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
+static hda_nid_t stac92hd83xxx_mux_nids[2] = {
0x17, 0x18,
};
@@ -362,9 +375,12 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = {
0x03, 0x0c, 0x20, 0x40,
};
-static hda_nid_t stac92hd83xxx_amp_nids[1] = {
- 0xc,
+#define STAC92HD83XXX_NUM_CAPS 2
+static unsigned long stac92hd83xxx_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_OUTPUT),
};
+#define stac92hd83xxx_capsws stac92hd83xxx_capvols
static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
0x0a, 0x0d, 0x0f
@@ -395,6 +411,13 @@ static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
0x22, 0
};
+#define STAC92HD71BXX_NUM_CAPS 2
+static unsigned long stac92hd71bxx_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+};
+#define stac92hd71bxx_capsws stac92hd71bxx_capvols
+
static hda_nid_t stac925x_adc_nids[1] = {
0x03,
};
@@ -416,6 +439,13 @@ static hda_nid_t stac925x_dmux_nids[1] = {
0x14,
};
+static unsigned long stac925x_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+};
+static unsigned long stac925x_capsws[] = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+};
+
static hda_nid_t stac922x_adc_nids[2] = {
0x06, 0x07,
};
@@ -424,6 +454,13 @@ static hda_nid_t stac922x_mux_nids[2] = {
0x12, 0x13,
};
+#define STAC922X_NUM_CAPS 2
+static unsigned long stac922x_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
+};
+#define stac922x_capsws stac922x_capvols
+
static hda_nid_t stac927x_slave_dig_outs[2] = {
0x1f, 0,
};
@@ -453,6 +490,18 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
0x13, 0x14, 0
};
+#define STAC927X_NUM_CAPS 3
+static unsigned long stac927x_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT),
+};
+static unsigned long stac927x_capsws[] = {
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+};
+
static const char *stac927x_spdif_labels[5] = {
"Digital Playback", "ADAT", "Analog Mux 1",
"Analog Mux 2", "Analog Mux 3"
@@ -479,6 +528,16 @@ static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
0x17, 0x18, 0
};
+#define STAC9205_NUM_CAPS 2
+static unsigned long stac9205_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT),
+};
+static unsigned long stac9205_capsws[] = {
+ HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT),
+};
+
static hda_nid_t stac9200_pin_nids[8] = {
0x08, 0x09, 0x0d, 0x0e,
0x0f, 0x10, 0x11, 0x12,
@@ -529,34 +588,6 @@ static hda_nid_t stac9205_pin_nids[12] = {
0x21, 0x22,
};
-#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info
-
-static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct sigmatel_spec *spec = codec->spec;
- hda_nid_t nid = spec->amp_nids[spec->cur_amux];
-
- kcontrol->private_value ^= get_amp_nid(kcontrol);
- kcontrol->private_value |= nid;
-
- return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
-}
-
-static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct sigmatel_spec *spec = codec->spec;
- hda_nid_t nid = spec->amp_nids[spec->cur_amux];
-
- kcontrol->private_value ^= get_amp_nid(kcontrol);
- kcontrol->private_value |= nid;
-
- return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
-}
-
static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -693,9 +724,35 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct sigmatel_spec *spec = codec->spec;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
- return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
+ const struct hda_input_mux *imux = spec->input_mux;
+ unsigned int idx, prev_idx;
+
+ idx = ucontrol->value.enumerated.item[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+ prev_idx = spec->cur_mux[adc_idx];
+ if (prev_idx == idx)
+ return 0;
+ if (idx < spec->num_analog_muxes) {
+ snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[idx].index);
+ if (prev_idx >= spec->num_analog_muxes) {
+ imux = spec->dinput_mux;
+ /* 0 = analog */
+ snd_hda_codec_write_cache(codec,
+ spec->dmux_nids[adc_idx], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[0].index);
+ }
+ } else {
+ imux = spec->dinput_mux;
+ snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[idx - 1].index);
+ }
+ spec->cur_mux[adc_idx] = idx;
+ return 1;
}
static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol,
@@ -726,41 +783,6 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol,
spec->mono_nid, &spec->cur_mmux);
}
-static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct sigmatel_spec *spec = codec->spec;
- return snd_hda_input_mux_info(spec->amp_mux, uinfo);
-}
-
-static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct sigmatel_spec *spec = codec->spec;
-
- ucontrol->value.enumerated.item[0] = spec->cur_amux;
- return 0;
-}
-
-static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct sigmatel_spec *spec = codec->spec;
- struct snd_kcontrol *ctl =
- snd_hda_find_mixer_ctl(codec, "Amp Capture Volume");
- if (!ctl)
- return -EINVAL;
-
- snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE |
- SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
-
- return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol,
- 0, &spec->cur_amux);
-}
-
#define stac92xx_aloopback_info snd_ctl_boolean_mono_info
static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol,
@@ -828,84 +850,16 @@ static struct hda_verb stac9200_eapd_init[] = {
{}
};
-static struct hda_verb stac92hd73xx_6ch_core_init[] = {
- /* set master volume and direct control */
- { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- /* setup adcs to point to mixer */
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* setup import muxs */
- { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {}
-};
-
static struct hda_verb dell_eq_core_init[] = {
/* set master volume to max value without distortion
* and direct control */
{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
- /* setup adcs to point to mixer */
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
- /* setup import muxs */
- { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {}
-};
-
-static struct hda_verb dell_m6_core_init[] = {
- { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- /* setup adcs to point to mixer */
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
- /* setup import muxs */
- { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {}
-};
-
-static struct hda_verb stac92hd73xx_8ch_core_init[] = {
- /* set master volume and direct control */
- { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- /* setup adcs to point to mixer */
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* setup import muxs */
- { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
{}
};
-static struct hda_verb stac92hd73xx_10ch_core_init[] = {
+static struct hda_verb stac92hd73xx_core_init[] = {
/* set master volume and direct control */
{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- /* dac3 is connected to import3 mux */
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f},
- /* setup adcs to point to mixer */
- { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* setup import muxs */
- { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
- { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
{}
};
@@ -925,19 +879,6 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
{}
};
-#define HD_DISABLE_PORTF 1
-static struct hda_verb stac92hd71bxx_analog_core_init[] = {
- /* start of config #1 */
-
- /* connect port 0f to audio mixer */
- { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
- /* start of config #2 */
-
- /* set master volume and direct control */
- { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
- {}
-};
-
static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
/* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -996,31 +937,6 @@ static struct hda_verb stac9205_core_init[] = {
.put = stac92xx_mono_mux_enum_put, \
}
-#define STAC_AMP_MUX \
- { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = "Amp Selector Capture Switch", \
- .count = 1, \
- .info = stac92xx_amp_mux_enum_info, \
- .get = stac92xx_amp_mux_enum_get, \
- .put = stac92xx_amp_mux_enum_put, \
- }
-
-#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \
- { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .index = 0, \
- .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
- SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
- .info = stac92xx_amp_volume_info, \
- .get = stac92xx_amp_volume_get, \
- .put = stac92xx_amp_volume_put, \
- .tlv = { .c = snd_hda_mixer_amp_tlv }, \
- .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \
- }
-
#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -1051,34 +967,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = {
{ } /* end */
};
-#define DELL_M6_MIXER 6
-static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
- /* start of config #1 */
- HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
-
- /* start of config #2 */
- HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
-
- { } /* end */
-};
-
static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
{}
@@ -1094,134 +982,14 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
{}
};
-static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
- { } /* end */
-};
-
-static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
-
- HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
-
- HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
- { } /* end */
-};
-
-
-static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT),
-
- HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT),
-
- /*
- HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT),
- */
- { } /* end */
-};
-
-static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
- /* analog pc-beep replaced with digital beep support */
- /*
- HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
- */
-
- HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
-
- HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
-
- HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
-
- HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
- HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
- { } /* end */
-};
static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
};
-static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
static struct snd_kcontrol_new stac925x_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
- { } /* end */
-};
-
-static struct snd_kcontrol_new stac9205_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -1230,29 +998,6 @@ static struct snd_kcontrol_new stac9205_loopback[] = {
{}
};
-/* This needs to be generated dynamically based on sequence */
-static struct snd_kcontrol_new stac922x_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-
-static struct snd_kcontrol_new stac927x_mixer[] = {
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
static struct snd_kcontrol_new stac927x_loopback[] = {
STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
{}
@@ -1310,16 +1055,19 @@ static int stac92xx_build_controls(struct hda_codec *codec)
int err;
int i;
- err = snd_hda_add_new_ctls(codec, spec->mixer);
- if (err < 0)
- return err;
+ if (spec->mixer) {
+ err = snd_hda_add_new_ctls(codec, spec->mixer);
+ if (err < 0)
+ return err;
+ }
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
return err;
}
- if (spec->num_dmuxes > 0) {
+ if (!spec->auto_mic && spec->num_dmuxes > 0 &&
+ snd_hda_get_bool_hint(codec, "separate_dmux") == 1) {
stac_dmux_mixer.count = spec->num_dmuxes;
err = snd_hda_ctl_add(codec,
snd_ctl_new1(&stac_dmux_mixer, codec));
@@ -1766,12 +1514,20 @@ static unsigned int dell_m6_pin_configs[13] = {
0x4f0000f0,
};
+static unsigned int alienware_m17x_pin_configs[13] = {
+ 0x0321101f, 0x0321101f, 0x03a11020, 0x03014020,
+ 0x90170110, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0,
+ 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
+ 0x904601b0,
+};
+
static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
[STAC_92HD73XX_REF] = ref92hd73xx_pin_configs,
[STAC_DELL_M6_AMIC] = dell_m6_pin_configs,
[STAC_DELL_M6_DMIC] = dell_m6_pin_configs,
[STAC_DELL_M6_BOTH] = dell_m6_pin_configs,
[STAC_DELL_EQ] = dell_m6_pin_configs,
+ [STAC_ALIENWARE_M17X] = alienware_m17x_pin_configs,
};
static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
@@ -1783,6 +1539,7 @@ static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
[STAC_DELL_M6_DMIC] = "dell-m6-dmic",
[STAC_DELL_M6_BOTH] = "dell-m6",
[STAC_DELL_EQ] = "dell-eq",
+ [STAC_ALIENWARE_M17X] = "alienware",
};
static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
@@ -1820,6 +1577,12 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
{} /* terminator */
};
+static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
+ "Alienware M17x", STAC_ALIENWARE_M17X),
+ {} /* terminator */
+};
+
static unsigned int ref92hd83xxx_pin_configs[10] = {
0x02214030, 0x02211010, 0x02a19020, 0x02170130,
0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
@@ -1927,6 +1690,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"HP mini 1000", STAC_HP_M4),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
"HP HDX", STAC_HP_HDX), /* HDX16 */
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
+ "HP", STAC_HP_DV5),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
"unknown Dell", STAC_DELL_M4_1),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
@@ -2642,8 +2407,7 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
return 0;
}
-static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
- unsigned char type);
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid);
static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2657,7 +2421,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
/* check to be sure that the ports are upto date with
* switch changes
*/
- stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
+ stac_issue_unsol_event(codec, nid);
return 1;
}
@@ -2790,7 +2554,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
* appropriately according to the pin direction
*/
if (spec->hp_detect)
- stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
+ stac_issue_unsol_event(codec, nid);
return 1;
}
@@ -2859,8 +2623,6 @@ enum {
STAC_CTL_WIDGET_VOL,
STAC_CTL_WIDGET_MUTE,
STAC_CTL_WIDGET_MONO_MUX,
- STAC_CTL_WIDGET_AMP_MUX,
- STAC_CTL_WIDGET_AMP_VOL,
STAC_CTL_WIDGET_HP_SWITCH,
STAC_CTL_WIDGET_IO_SWITCH,
STAC_CTL_WIDGET_CLFE_SWITCH,
@@ -2871,8 +2633,6 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
STAC_MONO_MUX,
- STAC_AMP_MUX,
- STAC_AMP_VOL(NULL, 0, 0, 0, 0),
STAC_CODEC_HP_SWITCH(NULL),
STAC_CODEC_IO_SWITCH(NULL, 0),
STAC_CODEC_CLFE_SWITCH(NULL, 0),
@@ -2973,6 +2733,8 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec)
struct snd_kcontrol_new *knew;
struct hda_input_mux *imux = &spec->private_imux;
+ if (spec->auto_mic)
+ return 0; /* no need for input source */
if (!spec->num_adcs || imux->num_items <= 1)
return 0; /* no need for input source control */
knew = stac_control_new(spec, &stac_input_src_temp,
@@ -3066,7 +2828,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
HDA_MAX_CONNECTIONS);
for (j = 0; j < conn_len; j++) {
wcaps = get_wcaps(codec, conn[j]);
- wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ wtype = get_wcaps_type(wcaps);
/* we check only analog outputs */
if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL))
continue;
@@ -3325,6 +3087,21 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
return 0;
}
+static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol,
+ unsigned long sw, int idx)
+{
+ int err;
+ err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx,
+ "Capture Volume", vol);
+ if (err < 0)
+ return err;
+ err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_MUTE, idx,
+ "Capture Switch", sw);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
/* add playback controls from the parsed DAC table */
static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
@@ -3398,7 +3175,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
spec->mono_nid,
con_lst,
HDA_MAX_NUM_INPUTS);
- if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_mono_labels))
+ if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels))
return -EINVAL;
for (i = 0; i < num_cons; i++) {
@@ -3412,37 +3189,6 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
"Mono Mux", spec->mono_nid);
}
-/* labels for amp mux outputs */
-static const char *stac92xx_amp_labels[3] = {
- "Front Microphone", "Microphone", "Line In",
-};
-
-/* create amp out controls mux on capable codecs */
-static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
- struct hda_input_mux *amp_mux = &spec->private_amp_mux;
- int i, err;
-
- for (i = 0; i < spec->num_amps; i++) {
- amp_mux->items[amp_mux->num_items].label =
- stac92xx_amp_labels[i];
- amp_mux->items[amp_mux->num_items].index = i;
- amp_mux->num_items++;
- }
-
- if (spec->num_amps > 1) {
- err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX,
- "Amp Selector Capture Switch", 0);
- if (err < 0)
- return err;
- }
- return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL,
- "Amp Capture Volume",
- HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT));
-}
-
-
/* create PC beep volume controls */
static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
hda_nid_t nid)
@@ -3511,19 +3257,33 @@ static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
- int wcaps, nid, i, err = 0;
+ int i, j, err = 0;
for (i = 0; i < spec->num_muxes; i++) {
+ hda_nid_t nid;
+ unsigned int wcaps;
+ unsigned long val;
+
nid = spec->mux_nids[i];
wcaps = get_wcaps(codec, nid);
+ if (!(wcaps & AC_WCAP_OUT_AMP))
+ continue;
- if (wcaps & AC_WCAP_OUT_AMP) {
- err = stac92xx_add_control_idx(spec,
- STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume",
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
+ /* check whether already the same control was created as
+ * normal Capture Volume.
+ */
+ val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ for (j = 0; j < spec->num_caps; j++) {
+ if (spec->capvols[j] == val)
+ break;
}
+ if (j < spec->num_caps)
+ continue;
+
+ err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, i,
+ "Mux Capture Volume", val);
+ if (err < 0)
+ return err;
}
return 0;
};
@@ -3544,7 +3304,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
spec->smux_nids[0],
con_lst,
HDA_MAX_NUM_INPUTS);
- if (!num_cons)
+ if (num_cons <= 0)
return -EINVAL;
if (!labels)
@@ -3565,101 +3325,231 @@ static const char *stac92xx_dmic_labels[5] = {
"Digital Mic 3", "Digital Mic 4"
};
+static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
+ hda_nid_t nid)
+{
+ hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+ int i, nums;
+
+ nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+ for (i = 0; i < nums; i++)
+ if (conn[i] == nid)
+ return i;
+ return -1;
+}
+
+/* create a volume assigned to the given pin (only if supported) */
+/* return 1 if the volume control is created */
+static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid,
+ const char *label, int direction)
+{
+ unsigned int caps, nums;
+ char name[32];
+ int err;
+
+ if (direction == HDA_OUTPUT)
+ caps = AC_WCAP_OUT_AMP;
+ else
+ caps = AC_WCAP_IN_AMP;
+ if (!(get_wcaps(codec, nid) & caps))
+ return 0;
+ caps = query_amp_caps(codec, nid, direction);
+ nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ if (!nums)
+ return 0;
+ snprintf(name, sizeof(name), "%s Capture Volume", label);
+ err = stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction));
+ if (err < 0)
+ return err;
+ return 1;
+}
+
/* create playback/capture controls for input pins on dmic capable codecs */
static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *imux = &spec->private_imux;
struct hda_input_mux *dimux = &spec->private_dimux;
- hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
- int err, i, j;
- char name[32];
+ int err, i, active_mics;
+ unsigned int def_conf;
dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0];
dimux->items[dimux->num_items].index = 0;
dimux->num_items++;
+ active_mics = 0;
+ for (i = 0; i < spec->num_dmics; i++) {
+ /* check the validity: sometimes it's a dead vendor-spec node */
+ if (get_wcaps_type(get_wcaps(codec, spec->dmic_nids[i]))
+ != AC_WID_PIN)
+ continue;
+ def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
+ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
+ active_mics++;
+ }
+
for (i = 0; i < spec->num_dmics; i++) {
hda_nid_t nid;
int index;
- int num_cons;
- unsigned int wcaps;
- unsigned int def_conf;
+ const char *label;
- def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
+ nid = spec->dmic_nids[i];
+ if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+ continue;
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
continue;
- nid = spec->dmic_nids[i];
- num_cons = snd_hda_get_connections(codec,
- spec->dmux_nids[0],
- con_lst,
- HDA_MAX_NUM_INPUTS);
- for (j = 0; j < num_cons; j++)
- if (con_lst[j] == nid) {
- index = j;
- goto found;
- }
- continue;
-found:
- wcaps = get_wcaps(codec, nid) &
- (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
+ index = get_connection_index(codec, spec->dmux_nids[0], nid);
+ if (index < 0)
+ continue;
- if (wcaps) {
- sprintf(name, "%s Capture Volume",
- stac92xx_dmic_labels[dimux->num_items]);
+ if (active_mics == 1)
+ label = "Digital Mic";
+ else
+ label = stac92xx_dmic_labels[dimux->num_items];
- err = stac92xx_add_control(spec,
- STAC_CTL_WIDGET_VOL,
- name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- (wcaps & AC_WCAP_OUT_AMP) ?
- HDA_OUTPUT : HDA_INPUT));
+ err = create_elem_capture_vol(codec, nid, label, HDA_INPUT);
+ if (err < 0)
+ return err;
+ if (!err) {
+ err = create_elem_capture_vol(codec, nid, label,
+ HDA_OUTPUT);
if (err < 0)
return err;
}
- dimux->items[dimux->num_items].label =
- stac92xx_dmic_labels[dimux->num_items];
+ dimux->items[dimux->num_items].label = label;
dimux->items[dimux->num_items].index = index;
dimux->num_items++;
+ if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+ imux->items[imux->num_items].label = label;
+ imux->items[imux->num_items].index = index;
+ imux->num_items++;
+ }
}
return 0;
}
+static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *fixed, hda_nid_t *ext)
+{
+ unsigned int cfg;
+
+ if (!nid)
+ return 0;
+ cfg = snd_hda_codec_get_pincfg(codec, nid);
+ switch (get_defcfg_connect(cfg)) {
+ case AC_JACK_PORT_FIXED:
+ if (*fixed)
+ return 1; /* already occupied */
+ *fixed = nid;
+ break;
+ case AC_JACK_PORT_COMPLEX:
+ if (*ext)
+ return 1; /* already occupied */
+ *ext = nid;
+ break;
+ }
+ return 0;
+}
+
+static int set_mic_route(struct hda_codec *codec,
+ struct sigmatel_mic_route *mic,
+ hda_nid_t pin)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ mic->pin = pin;
+ for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++)
+ if (pin == cfg->input_pins[i])
+ break;
+ if (i <= AUTO_PIN_FRONT_MIC) {
+ /* analog pin */
+ mic->dmux_idx = 0;
+ i = get_connection_index(codec, spec->mux_nids[0], pin);
+ if (i < 0)
+ return -1;
+ mic->mux_idx = i;
+ } else if (spec->dmux_nids) {
+ /* digital pin */
+ mic->mux_idx = 0;
+ i = get_connection_index(codec, spec->dmux_nids[0], pin);
+ if (i < 0)
+ return -1;
+ mic->dmux_idx = i;
+ }
+ return 0;
+}
+
+/* return non-zero if the device is for automatic mic switch */
+static int stac_check_auto_mic(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t fixed, ext;
+ int i;
+
+ for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) {
+ if (cfg->input_pins[i])
+ return 0; /* must be exclusively mics */
+ }
+ fixed = ext = 0;
+ for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++)
+ if (check_mic_pin(codec, cfg->input_pins[i], &fixed, &ext))
+ return 0;
+ for (i = 0; i < spec->num_dmics; i++)
+ if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext))
+ return 0;
+ if (!fixed || !ext)
+ return 0;
+ if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+ return 0; /* no unsol support */
+ if (set_mic_route(codec, &spec->ext_mic, ext) ||
+ set_mic_route(codec, &spec->int_mic, fixed))
+ return 0; /* something is wrong */
+ return 1;
+}
+
/* create playback/capture controls for input pins */
static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
{
struct sigmatel_spec *spec = codec->spec;
struct hda_input_mux *imux = &spec->private_imux;
- hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
- int i, j, k;
+ int i, j;
for (i = 0; i < AUTO_PIN_LAST; i++) {
- int index;
+ hda_nid_t nid = cfg->input_pins[i];
+ int index, err;
- if (!cfg->input_pins[i])
+ if (!nid)
continue;
index = -1;
for (j = 0; j < spec->num_muxes; j++) {
- int num_cons;
- num_cons = snd_hda_get_connections(codec,
- spec->mux_nids[j],
- con_lst,
- HDA_MAX_NUM_INPUTS);
- for (k = 0; k < num_cons; k++)
- if (con_lst[k] == cfg->input_pins[i]) {
- index = k;
- goto found;
- }
+ index = get_connection_index(codec, spec->mux_nids[j],
+ nid);
+ if (index >= 0)
+ break;
}
- continue;
- found:
+ if (index < 0)
+ continue;
+
+ err = create_elem_capture_vol(codec, nid,
+ auto_pin_cfg_labels[i],
+ HDA_INPUT);
+ if (err < 0)
+ return err;
+
imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
imux->items[imux->num_items].index = index;
imux->num_items++;
}
+ spec->num_analog_muxes = imux->num_items;
if (imux->num_items) {
/*
@@ -3711,7 +3601,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
{
struct sigmatel_spec *spec = codec->spec;
int hp_swap = 0;
- int err;
+ int i, err;
if ((err = snd_hda_parse_pin_def_config(codec,
&spec->autocfg,
@@ -3751,11 +3641,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
if (snd_hda_get_connections(codec,
spec->autocfg.mono_out_pin, conn_list, 1) &&
snd_hda_get_connections(codec, conn_list[0],
- conn_list, 1)) {
+ conn_list, 1) > 0) {
int wcaps = get_wcaps(codec, conn_list[0]);
- int wid_type = (wcaps & AC_WCAP_TYPE)
- >> AC_WCAP_TYPE_SHIFT;
+ int wid_type = get_wcaps_type(wcaps);
/* LR swap check, some stac925x have a mux that
* changes the DACs output path instead of the
* mono-mux path.
@@ -3846,6 +3735,21 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
spec->autocfg.line_outs = 0;
}
+ if (stac_check_auto_mic(codec)) {
+ spec->auto_mic = 1;
+ /* only one capture for auto-mic */
+ spec->num_adcs = 1;
+ spec->num_caps = 1;
+ spec->num_muxes = 1;
+ }
+
+ for (i = 0; i < spec->num_caps; i++) {
+ err = stac92xx_add_capvol_ctls(codec, spec->capvols[i],
+ spec->capsws[i], i);
+ if (err < 0)
+ return err;
+ }
+
err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
@@ -3855,11 +3759,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
if (err < 0)
return err;
}
- if (spec->num_amps > 0) {
- err = stac92xx_auto_create_amp_output_ctls(codec);
- if (err < 0)
- return err;
- }
if (spec->num_dmics > 0 && !spec->dinput_mux)
if ((err = stac92xx_auto_create_dmic_input_ctls(codec,
&spec->autocfg)) < 0)
@@ -3896,7 +3795,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
spec->dinput_mux = &spec->private_dimux;
spec->sinput_mux = &spec->private_smux;
spec->mono_mux = &spec->private_mono_mux;
- spec->amp_mux = &spec->private_amp_mux;
return 1;
}
@@ -4108,14 +4006,14 @@ static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
}
static struct sigmatel_event *stac_get_event(struct hda_codec *codec,
- hda_nid_t nid, unsigned char type)
+ hda_nid_t nid)
{
struct sigmatel_spec *spec = codec->spec;
struct sigmatel_event *event = spec->events.list;
int i;
for (i = 0; i < spec->events.used; i++, event++) {
- if (event->nid == nid && event->type == type)
+ if (event->nid == nid)
return event;
}
return NULL;
@@ -4135,24 +4033,32 @@ static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec,
return NULL;
}
-static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
- unsigned int type)
+/* check if given nid is a valid pin and no other events are assigned
+ * to it. If OK, assign the event, set the unsol flag, and returns 1.
+ * Otherwise, returns zero.
+ */
+static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int type)
{
struct sigmatel_event *event;
int tag;
if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
- return;
- event = stac_get_event(codec, nid, type);
- if (event)
+ return 0;
+ event = stac_get_event(codec, nid);
+ if (event) {
+ if (event->type != type)
+ return 0;
tag = event->tag;
- else
+ } else {
tag = stac_add_event(codec->spec, nid, type, 0);
- if (tag < 0)
- return;
+ if (tag < 0)
+ return 0;
+ }
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_UNSOLICITED_ENABLE,
AC_USRSP_EN | tag);
+ return 1;
}
static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
@@ -4245,20 +4151,36 @@ static int stac92xx_init(struct hda_codec *codec)
hda_nid_t nid = cfg->hp_pins[i];
enable_pin_detect(codec, nid, STAC_HP_EVENT);
}
+ if (cfg->line_out_type == AUTO_PIN_LINE_OUT &&
+ cfg->speaker_outs > 0) {
+ /* enable pin-detect for line-outs as well */
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t nid = cfg->line_out_pins[i];
+ enable_pin_detect(codec, nid, STAC_LO_EVENT);
+ }
+ }
+
/* force to enable the first line-out; the others are set up
* in unsol_event
*/
stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
AC_PINCTL_OUT_EN);
/* fake event to set up pins */
- stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
- STAC_HP_EVENT);
+ stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
} else {
stac92xx_auto_init_multi_out(codec);
stac92xx_auto_init_hp_out(codec);
for (i = 0; i < cfg->hp_outs; i++)
stac_toggle_power_map(codec, cfg->hp_pins[i], 1);
}
+ if (spec->auto_mic) {
+ /* initialize connection to analog input */
+ if (spec->dmux_nids)
+ snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
+ AC_VERB_SET_CONNECT_SEL, 0);
+ if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT))
+ stac_issue_unsol_event(codec, spec->ext_mic.pin);
+ }
for (i = 0; i < AUTO_PIN_LAST; i++) {
hda_nid_t nid = cfg->input_pins[i];
if (nid) {
@@ -4285,10 +4207,9 @@ static int stac92xx_init(struct hda_codec *codec)
}
conf = snd_hda_codec_get_pincfg(codec, nid);
if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) {
- enable_pin_detect(codec, nid,
- STAC_INSERT_EVENT);
- stac_issue_unsol_event(codec, nid,
- STAC_INSERT_EVENT);
+ if (enable_pin_detect(codec, nid,
+ STAC_INSERT_EVENT))
+ stac_issue_unsol_event(codec, nid);
}
}
}
@@ -4333,10 +4254,8 @@ static int stac92xx_init(struct hda_codec *codec)
stac_toggle_power_map(codec, nid, 1);
continue;
}
- if (!stac_get_event(codec, nid, STAC_INSERT_EVENT)) {
- enable_pin_detect(codec, nid, STAC_PWR_EVENT);
- stac_issue_unsol_event(codec, nid, STAC_PWR_EVENT);
- }
+ if (enable_pin_detect(codec, nid, STAC_PWR_EVENT))
+ stac_issue_unsol_event(codec, nid);
}
if (spec->dac_list)
stac92xx_power_down(codec);
@@ -4440,6 +4359,48 @@ static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
return 0;
}
+static void stac92xx_line_out_detect(struct hda_codec *codec,
+ int presence)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (presence)
+ break;
+ presence = get_pin_presence(codec, cfg->line_out_pins[i]);
+ if (presence) {
+ unsigned int pinctl;
+ pinctl = snd_hda_codec_read(codec,
+ cfg->line_out_pins[i], 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (pinctl & AC_PINCTL_IN_EN)
+ presence = 0; /* mic- or line-input */
+ }
+ }
+
+ if (presence) {
+ /* disable speakers */
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask && spec->eapd_switch)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
+ } else {
+ /* enable speakers */
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask && spec->eapd_switch)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data |
+ spec->eapd_mask);
+ }
+}
+
/* return non-zero if the hp-pin of the given array index isn't
* a jack-detection target
*/
@@ -4492,13 +4453,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
for (i = 0; i < cfg->line_outs; i++)
stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
- for (i = 0; i < cfg->speaker_outs; i++)
- stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
- AC_PINCTL_OUT_EN);
- if (spec->eapd_mask && spec->eapd_switch)
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data &
- ~spec->eapd_mask);
} else {
/* enable lineouts */
if (spec->hp_switch)
@@ -4507,14 +4461,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
for (i = 0; i < cfg->line_outs; i++)
stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
AC_PINCTL_OUT_EN);
- for (i = 0; i < cfg->speaker_outs; i++)
- stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
- AC_PINCTL_OUT_EN);
- if (spec->eapd_mask && spec->eapd_switch)
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data |
- spec->eapd_mask);
}
+ stac92xx_line_out_detect(codec, presence);
/* toggle hp outs */
for (i = 0; i < cfg->hp_outs; i++) {
unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
@@ -4599,10 +4547,28 @@ static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid)
}
}
-static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
- unsigned char type)
+static void stac92xx_mic_detect(struct hda_codec *codec)
{
- struct sigmatel_event *event = stac_get_event(codec, nid, type);
+ struct sigmatel_spec *spec = codec->spec;
+ struct sigmatel_mic_route *mic;
+
+ if (get_pin_presence(codec, spec->ext_mic.pin))
+ mic = &spec->ext_mic;
+ else
+ mic = &spec->int_mic;
+ if (mic->dmux_idx)
+ snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ mic->dmux_idx);
+ else
+ snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ mic->mux_idx);
+}
+
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct sigmatel_event *event = stac_get_event(codec, nid);
if (!event)
return;
codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26);
@@ -4621,8 +4587,18 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
switch (event->type) {
case STAC_HP_EVENT:
+ case STAC_LO_EVENT:
stac92xx_hp_detect(codec);
- /* fallthru */
+ break;
+ case STAC_MIC_EVENT:
+ stac92xx_mic_detect(codec);
+ break;
+ }
+
+ switch (event->type) {
+ case STAC_HP_EVENT:
+ case STAC_LO_EVENT:
+ case STAC_MIC_EVENT:
case STAC_INSERT_EVENT:
case STAC_PWR_EVENT:
if (spec->num_pwrs > 0)
@@ -4713,8 +4689,7 @@ static int stac92xx_resume(struct hda_codec *codec)
snd_hda_codec_resume_cache(codec);
/* fake event to set up pins again to override cached values */
if (spec->hp_detect)
- stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
- STAC_HP_EVENT);
+ stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
return 0;
}
@@ -4754,6 +4729,19 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
{
struct sigmatel_spec *spec = codec->spec;
+ int i;
+ hda_nid_t nid;
+
+ /* reset each pin before powering down DAC/ADC to avoid click noise */
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int wcaps = get_wcaps(codec, nid);
+ unsigned int wid_type = get_wcaps_type(wcaps);
+ if (wid_type == AC_WID_PIN)
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ }
+
if (spec->eapd_mask)
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data &
@@ -4790,7 +4778,8 @@ static int patch_stac9200(struct hda_codec *codec)
stac9200_models,
stac9200_cfg_tbl);
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac9200_brd_tbl[spec->board_config]);
@@ -4862,8 +4851,8 @@ static int patch_stac925x(struct hda_codec *codec)
stac925x_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x,"
- "using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac925x_brd_tbl[spec->board_config]);
@@ -4893,6 +4882,9 @@ static int patch_stac925x(struct hda_codec *codec)
spec->init = stac925x_core_init;
spec->mixer = stac925x_mixer;
+ spec->num_caps = 1;
+ spec->capvols = stac925x_capvols;
+ spec->capsws = stac925x_capsws;
err = stac92xx_parse_auto_config(codec, 0x8, 0x7);
if (!err) {
@@ -4914,16 +4906,6 @@ static int patch_stac925x(struct hda_codec *codec)
return 0;
}
-static struct hda_input_mux stac92hd73xx_dmux = {
- .num_items = 4,
- .items = {
- { "Analog Inputs", 0x0b },
- { "Digital Mic 1", 0x09 },
- { "Digital Mic 2", 0x0a },
- { "CD", 0x08 },
- }
-};
-
static int patch_stac92hd73xx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
@@ -4943,10 +4925,16 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
STAC_92HD73XX_MODELS,
stac92hd73xx_models,
stac92hd73xx_cfg_tbl);
+ /* check codec subsystem id if not found */
+ if (spec->board_config < 0)
+ spec->board_config =
+ snd_hda_check_board_codec_sid_config(codec,
+ STAC_92HD73XX_MODELS, stac92hd73xx_models,
+ stac92hd73xx_codec_id_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for"
- " STAC92HD73XX, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac92hd73xx_brd_tbl[spec->board_config]);
@@ -4959,20 +4947,15 @@ again:
"number of channels defaulting to DAC count\n");
num_dacs = STAC92HD73_DAC_COUNT;
}
+ spec->init = stac92hd73xx_core_init;
switch (num_dacs) {
case 0x3: /* 6 Channel */
- spec->mixer = stac92hd73xx_6ch_mixer;
- spec->init = stac92hd73xx_6ch_core_init;
spec->aloopback_ctl = stac92hd73xx_6ch_loopback;
break;
case 0x4: /* 8 Channel */
- spec->mixer = stac92hd73xx_8ch_mixer;
- spec->init = stac92hd73xx_8ch_core_init;
spec->aloopback_ctl = stac92hd73xx_8ch_loopback;
break;
case 0x5: /* 10 Channel */
- spec->mixer = stac92hd73xx_10ch_mixer;
- spec->init = stac92hd73xx_10ch_core_init;
spec->aloopback_ctl = stac92hd73xx_10ch_loopback;
break;
}
@@ -4987,14 +4970,14 @@ again:
spec->dmic_nids = stac92hd73xx_dmic_nids;
spec->dmux_nids = stac92hd73xx_dmux_nids;
spec->smux_nids = stac92hd73xx_smux_nids;
- spec->amp_nids = stac92hd73xx_amp_nids;
- spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids);
spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids);
spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);
spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids);
- memcpy(&spec->private_dimux, &stac92hd73xx_dmux,
- sizeof(stac92hd73xx_dmux));
+
+ spec->num_caps = STAC92HD73XX_NUM_CAPS;
+ spec->capvols = stac92hd73xx_capvols;
+ spec->capsws = stac92hd73xx_capsws;
switch (spec->board_config) {
case STAC_DELL_EQ:
@@ -5004,43 +4987,40 @@ again:
case STAC_DELL_M6_DMIC:
case STAC_DELL_M6_BOTH:
spec->num_smuxes = 0;
- spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
- spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
spec->eapd_switch = 0;
- spec->num_amps = 1;
- if (spec->board_config != STAC_DELL_EQ)
- spec->init = dell_m6_core_init;
switch (spec->board_config) {
case STAC_DELL_M6_AMIC: /* Analog Mics */
snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
spec->num_dmics = 0;
- spec->private_dimux.num_items = 1;
break;
case STAC_DELL_M6_DMIC: /* Digital Mics */
snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
spec->num_dmics = 1;
- spec->private_dimux.num_items = 2;
break;
case STAC_DELL_M6_BOTH: /* Both */
snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
spec->num_dmics = 1;
- spec->private_dimux.num_items = 2;
break;
}
break;
+ case STAC_ALIENWARE_M17X:
+ spec->num_dmics = STAC92HD73XX_NUM_DMICS;
+ spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
+ spec->eapd_switch = 0;
+ break;
default:
spec->num_dmics = STAC92HD73XX_NUM_DMICS;
spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
spec->eapd_switch = 1;
+ break;
}
if (spec->board_config > STAC_92HD73XX_REF) {
/* GPIO0 High = Enable EAPD */
spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
spec->gpio_data = 0x01;
}
- spec->dinput_mux = &spec->private_dimux;
spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
spec->pwr_nids = stac92hd73xx_pwr_nids;
@@ -5072,15 +5052,6 @@ again:
return 0;
}
-static struct hda_input_mux stac92hd83xxx_dmux = {
- .num_items = 3,
- .items = {
- { "Analog Inputs", 0x03 },
- { "Digital Mic 1", 0x04 },
- { "Digital Mic 2", 0x05 },
- }
-};
-
static int patch_stac92hd83xxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
@@ -5097,32 +5068,30 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
spec->mono_nid = 0x19;
spec->digbeep_nid = 0x21;
- spec->dmic_nids = stac92hd83xxx_dmic_nids;
- spec->dmux_nids = stac92hd83xxx_dmux_nids;
+ spec->mux_nids = stac92hd83xxx_mux_nids;
+ spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
spec->adc_nids = stac92hd83xxx_adc_nids;
+ spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
spec->pwr_nids = stac92hd83xxx_pwr_nids;
- spec->amp_nids = stac92hd83xxx_amp_nids;
spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
spec->multiout.dac_nids = spec->dac_nids;
spec->init = stac92hd83xxx_core_init;
- spec->mixer = stac92hd83xxx_mixer;
spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
- spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
- spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
- spec->num_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids);
- spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
- spec->dinput_mux = &stac92hd83xxx_dmux;
spec->pin_nids = stac92hd83xxx_pin_nids;
+ spec->num_caps = STAC92HD83XXX_NUM_CAPS;
+ spec->capvols = stac92hd83xxx_capvols;
+ spec->capsws = stac92hd83xxx_capsws;
+
spec->board_config = snd_hda_check_board_config(codec,
STAC_92HD83XXX_MODELS,
stac92hd83xxx_models,
stac92hd83xxx_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for"
- " STAC92HD83XXX, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac92hd83xxx_brd_tbl[spec->board_config]);
@@ -5164,6 +5133,8 @@ again:
num_dacs = snd_hda_get_connections(codec, nid,
conn, STAC92HD83_DAC_COUNT + 1) - 1;
+ if (num_dacs < 0)
+ num_dacs = STAC92HD83_DAC_COUNT;
/* set port X to select the last DAC
*/
@@ -5177,25 +5148,6 @@ again:
return 0;
}
-static struct hda_input_mux stac92hd71bxx_dmux_nomixer = {
- .num_items = 3,
- .items = {
- { "Analog Inputs", 0x00 },
- { "Digital Mic 1", 0x02 },
- { "Digital Mic 2", 0x03 },
- }
-};
-
-static struct hda_input_mux stac92hd71bxx_dmux_amixer = {
- .num_items = 4,
- .items = {
- { "Analog Inputs", 0x00 },
- { "Mixer", 0x01 },
- { "Digital Mic 1", 0x02 },
- { "Digital Mic 2", 0x03 },
- }
-};
-
/* get the pin connection (fixed, none, etc) */
static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx)
{
@@ -5256,7 +5208,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
struct sigmatel_spec *spec;
struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
int err = 0;
- unsigned int ndmic_nids = 0;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -5285,8 +5236,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
stac92hd71bxx_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for"
- " STAC92HD71BXX, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac92hd71bxx_brd_tbl[spec->board_config]);
@@ -5301,6 +5252,10 @@ again:
spec->dmic_nids = stac92hd71bxx_dmic_nids;
spec->dmux_nids = stac92hd71bxx_dmux_nids;
+ spec->num_caps = STAC92HD71BXX_NUM_CAPS;
+ spec->capvols = stac92hd71bxx_capvols;
+ spec->capsws = stac92hd71bxx_capsws;
+
switch (codec->vendor_id) {
case 0x111d76b6: /* 4 Port without Analog Mixer */
case 0x111d76b7:
@@ -5308,24 +5263,13 @@ again:
/* fallthru */
case 0x111d76b4: /* 6 Port without Analog Mixer */
case 0x111d76b5:
- memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
- sizeof(stac92hd71bxx_dmux_nomixer));
- spec->mixer = stac92hd71bxx_mixer;
spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92hd71bxx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
STAC92HD71BXX_NUM_DMICS);
- if (spec->num_dmics) {
- spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
- spec->dinput_mux = &spec->private_dimux;
- ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
- }
break;
case 0x111d7608: /* 5 Port with Analog Mixer */
- memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
- sizeof(stac92hd71bxx_dmux_amixer));
- spec->private_dimux.num_items--;
switch (spec->board_config) {
case STAC_HP_M4:
/* Enable VREF power saving on GPIO1 detect */
@@ -5347,11 +5291,8 @@ again:
/* no output amps */
spec->num_pwrs = 0;
- spec->mixer = stac92hd71bxx_analog_mixer;
- spec->dinput_mux = &spec->private_dimux;
-
/* disable VSW */
- spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
+ spec->init = stac92hd71bxx_core_init;
unmute_init++;
snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5359,8 +5300,6 @@ again:
spec->num_dmics = stac92hd71bxx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
STAC92HD71BXX_NUM_DMICS - 1);
- spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
- ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
break;
case 0x111d7603: /* 6 Port with Analog Mixer */
if ((codec->revision_id & 0xf) == 1)
@@ -5370,17 +5309,12 @@ again:
spec->num_pwrs = 0;
/* fallthru */
default:
- memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
- sizeof(stac92hd71bxx_dmux_amixer));
- spec->dinput_mux = &spec->private_dimux;
- spec->mixer = stac92hd71bxx_analog_mixer;
- spec->init = stac92hd71bxx_analog_core_init;
+ spec->init = stac92hd71bxx_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
spec->num_dmics = stac92hd71bxx_connected_ports(codec,
stac92hd71bxx_dmic_nids,
STAC92HD71BXX_NUM_DMICS);
- spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
- ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
+ break;
}
if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
@@ -5408,6 +5342,7 @@ again:
spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
+ spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
switch (spec->board_config) {
@@ -5462,8 +5397,6 @@ again:
#endif
spec->multiout.dac_nids = spec->dac_nids;
- if (spec->dinput_mux)
- spec->private_dimux.num_items += spec->num_dmics - ndmic_nids;
err = stac92xx_parse_auto_config(codec, 0x21, 0);
if (!err) {
@@ -5541,8 +5474,8 @@ static int patch_stac922x(struct hda_codec *codec)
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
- "using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac922x_brd_tbl[spec->board_config]);
@@ -5555,7 +5488,10 @@ static int patch_stac922x(struct hda_codec *codec)
spec->num_pwrs = 0;
spec->init = stac922x_core_init;
- spec->mixer = stac922x_mixer;
+
+ spec->num_caps = STAC922X_NUM_CAPS;
+ spec->capvols = stac922x_capvols;
+ spec->capsws = stac922x_capsws;
spec->multiout.dac_nids = spec->dac_nids;
@@ -5604,8 +5540,8 @@ static int patch_stac927x(struct hda_codec *codec)
stac927x_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for"
- "STAC927x, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac927x_brd_tbl[spec->board_config]);
@@ -5630,7 +5566,6 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_dmics = 0;
spec->init = d965_core_init;
- spec->mixer = stac927x_mixer;
break;
case STAC_DELL_BIOS:
switch (codec->subsystem_id) {
@@ -5662,7 +5597,6 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_dmics = STAC927X_NUM_DMICS;
spec->init = d965_core_init;
- spec->mixer = stac927x_mixer;
spec->dmux_nids = stac927x_dmux_nids;
spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
break;
@@ -5675,9 +5609,12 @@ static int patch_stac927x(struct hda_codec *codec)
spec->num_dmics = 0;
spec->init = stac927x_core_init;
- spec->mixer = stac927x_mixer;
}
+ spec->num_caps = STAC927X_NUM_CAPS;
+ spec->capvols = stac927x_capvols;
+ spec->capsws = stac927x_capsws;
+
spec->num_pwrs = 0;
spec->aloopback_ctl = stac927x_loopback;
spec->aloopback_mask = 0x40;
@@ -5739,7 +5676,8 @@ static int patch_stac9205(struct hda_codec *codec)
stac9205_cfg_tbl);
again:
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac9205_brd_tbl[spec->board_config]);
@@ -5758,9 +5696,12 @@ static int patch_stac9205(struct hda_codec *codec)
spec->num_pwrs = 0;
spec->init = stac9205_core_init;
- spec->mixer = stac9205_mixer;
spec->aloopback_ctl = stac9205_loopback;
+ spec->num_caps = STAC9205_NUM_CAPS;
+ spec->capvols = stac9205_capvols;
+ spec->capsws = stac9205_capsws;
+
spec->aloopback_mask = 0x40;
spec->aloopback_shift = 0;
/* Turn on/off EAPD per HP plugging */
@@ -5835,12 +5776,6 @@ static struct hda_verb stac9872_core_init[] = {
{}
};
-static struct snd_kcontrol_new stac9872_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
- { } /* end */
-};
-
static hda_nid_t stac9872_pin_nids[] = {
0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f,
0x11, 0x13, 0x14,
@@ -5854,6 +5789,11 @@ static hda_nid_t stac9872_mux_nids[] = {
0x15
};
+static unsigned long stac9872_capvols[] = {
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+};
+#define stac9872_capsws stac9872_capvols
+
static unsigned int stac9872_vaio_pin_configs[9] = {
0x03211020, 0x411111f0, 0x411111f0, 0x03a15030,
0x411111f0, 0x90170110, 0x411111f0, 0x411111f0,
@@ -5891,8 +5831,8 @@ static int patch_stac9872(struct hda_codec *codec)
stac9872_models,
stac9872_cfg_tbl);
if (spec->board_config < 0)
- snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, "
- "using BIOS defaults\n");
+ snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+ codec->chip_name);
else
stac92xx_set_config_regs(codec,
stac9872_brd_tbl[spec->board_config]);
@@ -5902,8 +5842,10 @@ static int patch_stac9872(struct hda_codec *codec)
spec->adc_nids = stac9872_adc_nids;
spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
spec->mux_nids = stac9872_mux_nids;
- spec->mixer = stac9872_mixer;
spec->init = stac9872_core_init;
+ spec->num_caps = 1;
+ spec->capvols = stac9872_capvols;
+ spec->capsws = stac9872_capsws;
err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
if (err < 0) {
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e8f10b1..ee89db9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1339,8 +1339,7 @@ static int get_mux_nids(struct hda_codec *codec)
for (i = 0; i < spec->num_adc_nids; i++) {
nid = spec->adc_nids[i];
while (nid) {
- type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
- >> AC_WCAP_TYPE_SHIFT;
+ type = get_wcaps_type(get_wcaps(codec, nid));
if (type == AC_WID_PIN)
break;
n = snd_hda_get_connections(codec, nid, conn,
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index adc909e..9da2dae 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -379,6 +379,15 @@ struct snd_ice1712 {
unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate);
void (*set_spdif_clock)(struct snd_ice1712 *ice);
+#ifdef CONFIG_PM
+ int (*pm_suspend)(struct snd_ice1712 *);
+ int (*pm_resume)(struct snd_ice1712 *);
+ int pm_suspend_enabled:1;
+ int pm_saved_is_spdif_master:1;
+ unsigned int pm_saved_spdif_ctrl;
+ unsigned char pm_saved_spdif_cfg;
+ unsigned int pm_saved_route;
+#endif
};
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index cc84a83..af6e001 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -560,6 +560,7 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
spin_lock(&ice->reg_lock);
old = inb(ICEMT1724(ice, DMA_CONTROL));
if (cmd == SNDRV_PCM_TRIGGER_START)
@@ -570,6 +571,10 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
spin_unlock(&ice->reg_lock);
break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* apps will have to restart stream */
+ break;
+
default:
return -EINVAL;
}
@@ -2262,7 +2267,7 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice,
-static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice)
+static void snd_vt1724_chip_reset(struct snd_ice1712 *ice)
{
outb(VT1724_RESET , ICEREG1724(ice, CONTROL));
inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */
@@ -2272,7 +2277,7 @@ static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice)
msleep(10);
}
-static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice)
+static int snd_vt1724_chip_init(struct snd_ice1712 *ice)
{
outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG));
outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG));
@@ -2287,6 +2292,14 @@ static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice)
outb(0, ICEREG1724(ice, POWERDOWN));
+ /* MPU_RX and TX irq masks are cleared later dynamically */
+ outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK));
+
+ /* don't handle FIFO overrun/underruns (just yet),
+ * since they cause machine lockups
+ */
+ outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK));
+
return 0;
}
@@ -2431,6 +2444,8 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
snd_vt1724_proc_init(ice);
synchronize_irq(pci->irq);
+ card->private_data = ice;
+
err = pci_request_regions(pci, "ICE1724");
if (err < 0) {
kfree(ice);
@@ -2459,14 +2474,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card,
return -EIO;
}
- /* MPU_RX and TX irq masks are cleared later dynamically */
- outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK));
-
- /* don't handle FIFO overrun/underruns (just yet),
- * since they cause machine lockups
- */
- outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK));
-
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops);
if (err < 0) {
snd_vt1724_free(ice);
@@ -2650,11 +2657,96 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci)
pci_set_drvdata(pci, NULL);
}
+#ifdef CONFIG_PM
+static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ice1712 *ice = card->private_data;
+
+ if (!ice->pm_suspend_enabled)
+ return 0;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(ice->pcm);
+ snd_pcm_suspend_all(ice->pcm_pro);
+ snd_pcm_suspend_all(ice->pcm_ds);
+ snd_ac97_suspend(ice->ac97);
+
+ spin_lock_irq(&ice->reg_lock);
+ ice->pm_saved_is_spdif_master = ice->is_spdif_master(ice);
+ ice->pm_saved_spdif_ctrl = inw(ICEMT1724(ice, SPDIF_CTRL));
+ ice->pm_saved_spdif_cfg = inb(ICEREG1724(ice, SPDIF_CFG));
+ ice->pm_saved_route = inl(ICEMT1724(ice, ROUTE_PLAYBACK));
+ spin_unlock_irq(&ice->reg_lock);
+
+ if (ice->pm_suspend)
+ ice->pm_suspend(ice);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_vt1724_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ice1712 *ice = card->private_data;
+
+ if (!ice->pm_suspend_enabled)
+ return 0;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+
+ if (pci_enable_device(pci) < 0) {
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+
+ pci_set_master(pci);
+
+ snd_vt1724_chip_reset(ice);
+
+ if (snd_vt1724_chip_init(ice) < 0) {
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+
+ if (ice->pm_resume)
+ ice->pm_resume(ice);
+
+ if (ice->pm_saved_is_spdif_master) {
+ /* switching to external clock via SPDIF */
+ ice->set_spdif_clock(ice);
+ } else {
+ /* internal on-card clock */
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1);
+ }
+
+ update_spdif_bits(ice, ice->pm_saved_spdif_ctrl);
+
+ outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG));
+ outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK));
+
+ if (ice->ac97)
+ snd_ac97_resume(ice->ac97);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
static struct pci_driver driver = {
.name = "ICE1724",
.id_table = snd_vt1724_ids,
.probe = snd_vt1724_probe,
.remove = __devexit_p(snd_vt1724_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_vt1724_suspend,
+ .resume = snd_vt1724_resume,
+#endif
};
static int __init alsa_card_ice1724_init(void)
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 043a938..c75515f 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1077,7 +1077,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice)
/*
* initialize the chip
*/
-static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
+static void ak4396_init(struct snd_ice1712 *ice)
{
static unsigned short ak4396_inits[] = {
AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */
@@ -1087,9 +1087,37 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
AK4396_RCH_ATT, 0x00,
};
- struct prodigy_hifi_spec *spec;
unsigned int i;
+ /* initialize ak4396 codec */
+ /* reset codec */
+ ak4396_write(ice, AK4396_CTRL1, 0x86);
+ msleep(100);
+ ak4396_write(ice, AK4396_CTRL1, 0x87);
+
+ for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2)
+ ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]);
+}
+
+#ifdef CONFIG_PM
+static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice)
+{
+ /* initialize ak4396 codec and restore previous mixer volumes */
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i;
+ mutex_lock(&ice->gpio_mutex);
+ ak4396_init(ice);
+ for (i = 0; i < 2; i++)
+ ak4396_write(ice, AK4396_LCH_ATT + i, spec->vol[i] & 0xff);
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+#endif
+
+static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
+{
+ struct prodigy_hifi_spec *spec;
+
ice->vt1720 = 0;
ice->vt1724 = 1;
@@ -1112,14 +1140,12 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
return -ENOMEM;
ice->spec = spec;
- /* initialize ak4396 codec */
- /* reset codec */
- ak4396_write(ice, AK4396_CTRL1, 0x86);
- msleep(100);
- ak4396_write(ice, AK4396_CTRL1, 0x87);
-
- for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2)
- ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]);
+#ifdef CONFIG_PM
+ ice->pm_resume = &prodigy_hd2_resume;
+ ice->pm_suspend_enabled = 1;
+#endif
+
+ ak4396_init(ice);
return 0;
}
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index c1eb923..09b2b2a 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -215,17 +215,8 @@ EXPORT_SYMBOL(oxygen_write_spi);
void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data)
{
- unsigned long timeout;
-
/* should not need more than about 300 us */
- timeout = jiffies + msecs_to_jiffies(1);
- do {
- if (!(oxygen_read16(chip, OXYGEN_2WIRE_BUS_STATUS)
- & OXYGEN_2WIRE_BUSY))
- break;
- udelay(1);
- cond_resched();
- } while (time_after_eq(timeout, jiffies));
+ msleep(1);
oxygen_write8(chip, OXYGEN_2WIRE_MAP, map);
oxygen_write8(chip, OXYGEN_2WIRE_DATA, data);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 3da5c02..7bb827c 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3294,15 +3294,33 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
char *clock_source;
int x;
- if (hdsp_check_for_iobox (hdsp)) {
- snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n");
+ status = hdsp_read(hdsp, HDSP_statusRegister);
+ status2 = hdsp_read(hdsp, HDSP_status2Register);
+
+ snd_iprintf(buffer, "%s (Card #%d)\n", hdsp->card_name,
+ hdsp->card->number + 1);
+ snd_iprintf(buffer, "Buffers: capture %p playback %p\n",
+ hdsp->capture_buffer, hdsp->playback_buffer);
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdsp->irq, hdsp->port, (unsigned long)hdsp->iobase);
+ snd_iprintf(buffer, "Control register: 0x%x\n", hdsp->control_register);
+ snd_iprintf(buffer, "Control2 register: 0x%x\n",
+ hdsp->control2_register);
+ snd_iprintf(buffer, "Status register: 0x%x\n", status);
+ snd_iprintf(buffer, "Status2 register: 0x%x\n", status2);
+
+ if (hdsp_check_for_iobox(hdsp)) {
+ snd_iprintf(buffer, "No I/O box connected.\n"
+ "Please connect one and upload firmware.\n");
return;
- }
+ }
if (hdsp_check_for_firmware(hdsp, 0)) {
if (hdsp->state & HDSP_FirmwareCached) {
if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) {
- snd_iprintf(buffer, "Firmware loading from cache failed, please upload manually.\n");
+ snd_iprintf(buffer, "Firmware loading from "
+ "cache failed, "
+ "please upload manually.\n");
return;
}
} else {
@@ -3319,18 +3337,6 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
}
}
- status = hdsp_read(hdsp, HDSP_statusRegister);
- status2 = hdsp_read(hdsp, HDSP_status2Register);
-
- snd_iprintf(buffer, "%s (Card #%d)\n", hdsp->card_name, hdsp->card->number + 1);
- snd_iprintf(buffer, "Buffers: capture %p playback %p\n",
- hdsp->capture_buffer, hdsp->playback_buffer);
- snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
- hdsp->irq, hdsp->port, (unsigned long)hdsp->iobase);
- snd_iprintf(buffer, "Control register: 0x%x\n", hdsp->control_register);
- snd_iprintf(buffer, "Control2 register: 0x%x\n", hdsp->control2_register);
- snd_iprintf(buffer, "Status register: 0x%x\n", status);
- snd_iprintf(buffer, "Status2 register: 0x%x\n", status2);
snd_iprintf(buffer, "FIFO status: %d\n", hdsp_read(hdsp, HDSP_fifoStatus) & 0xff);
snd_iprintf(buffer, "MIDI1 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut0));
snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0));
@@ -3351,7 +3357,6 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
snd_iprintf(buffer, "\n");
-
switch (hdsp_clock_source(hdsp)) {
case HDSP_CLOCK_SOURCE_AUTOSYNC:
clock_source = "AutoSync";
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 2f09252..5518371 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -834,7 +834,7 @@ static irqreturn_t snd_ymfpci_interrupt(int irq, void *dev_id)
status = snd_ymfpci_readw(chip, YDSXGR_INTFLAG);
if (status & 1) {
if (chip->timer)
- snd_timer_interrupt(chip->timer, chip->timer->sticks);
+ snd_timer_interrupt(chip->timer, chip->timer_ticks);
}
snd_ymfpci_writew(chip, YDSXGR_INTFLAG, status);
@@ -1885,8 +1885,18 @@ static int snd_ymfpci_timer_start(struct snd_timer *timer)
unsigned int count;
chip = snd_timer_chip(timer);
- count = (timer->sticks << 1) - 1;
spin_lock_irqsave(&chip->reg_lock, flags);
+ if (timer->sticks > 1) {
+ chip->timer_ticks = timer->sticks;
+ count = timer->sticks - 1;
+ } else {
+ /*
+ * Divisor 1 is not allowed; fake it by using divisor 2 and
+ * counting two ticks for each interrupt.
+ */
+ chip->timer_ticks = 2;
+ count = 2 - 1;
+ }
snd_ymfpci_writew(chip, YDSXGR_TIMERCOUNT, count);
snd_ymfpci_writeb(chip, YDSXGR_TIMERCTRL, 0x03);
spin_unlock_irqrestore(&chip->reg_lock, flags);
@@ -1909,14 +1919,14 @@ static int snd_ymfpci_timer_precise_resolution(struct snd_timer *timer,
unsigned long *num, unsigned long *den)
{
*num = 1;
- *den = 48000;
+ *den = 96000;
return 0;
}
static struct snd_timer_hardware snd_ymfpci_timer_hw = {
.flags = SNDRV_TIMER_HW_AUTO,
- .resolution = 20833, /* 1/fs = 20.8333...us */
- .ticks = 0x8000,
+ .resolution = 10417, /* 1 / 96 kHz = 10.41666...us */
+ .ticks = 0x10000,
.start = snd_ymfpci_timer_start,
.stop = snd_ymfpci_timer_stop,
.precise_resolution = snd_ymfpci_timer_precise_resolution,
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index d3e786a..b1749bc 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -29,6 +29,7 @@ source "sound/soc/au1x/Kconfig"
source "sound/soc/blackfin/Kconfig"
source "sound/soc/davinci/Kconfig"
source "sound/soc/fsl/Kconfig"
+source "sound/soc/imx/Kconfig"
source "sound/soc/omap/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 6f1e28d..0c5eac0 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
@@ -7,6 +7,7 @@ obj-$(CONFIG_SND_SOC) += au1x/
obj-$(CONFIG_SND_SOC) += blackfin/
obj-$(CONFIG_SND_SOC) += davinci/
obj-$(CONFIG_SND_SOC) += fsl/
+obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += s3c24xx/
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 173a239..130b121 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -56,30 +56,14 @@
#define MCLK_RATE 12000000
-static struct clk *mclk;
-
-static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
- struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
- int ret;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
- MCLK_RATE, SND_SOC_CLOCK_IN);
- if (ret < 0) {
- clk_disable(mclk);
- return ret;
- }
-
- return 0;
-}
-
-static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+/*
+ * As shipped the board does not have inputs. However, it is relatively
+ * straightforward to modify the board to hook them up so support is left
+ * in the driver.
+ */
+#undef ENABLE_MIC_INPUT
- dev_dbg(rtd->socdev->dev, "shutdown");
-}
+static struct clk *mclk;
static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -87,102 +71,17 @@ static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- struct atmel_ssc_info *ssc_p = cpu_dai->private_data;
- struct ssc_device *ssc = ssc_p->ssc;
int ret;
- unsigned int rate;
- int cmr_div, period;
-
- if (ssc == NULL) {
- printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n");
- return -EINVAL;
- }
-
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /*
- * The SSC clock dividers depend on the sample rate. The CMR.DIV
- * field divides the system master clock MCK to drive the SSC TK
- * signal which provides the codec BCLK. The TCMR.PERIOD and
- * RCMR.PERIOD fields further divide the BCLK signal to drive
- * the SSC TF and RF signals which provide the codec DACLRC and
- * ADCLRC clocks.
- *
- * The dividers were determined through trial and error, where a
- * CMR.DIV value is chosen such that the resulting BCLK value is
- * divisible, or almost divisible, by (2 * sample rate), and then
- * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
- */
- rate = params_rate(params);
-
- switch (rate) {
- case 8000:
- cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */
- period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */
- break;
- case 11025:
- cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */
- period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */
- break;
- case 16000:
- cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */
- period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */
- break;
- case 22050:
- cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */
- period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */
- break;
- case 32000:
- cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */
- period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */
- break;
- case 44100:
- cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */
- period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */
- break;
- case 48000:
- cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */
- period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */
- break;
- case 88200:
- cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */
- period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */
- break;
- case 96000:
- cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */
- period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */
- break;
- default:
- printk(KERN_WARNING "unsupported rate %d"
- " on at91sam9g20ek board\n", rate);
- return -EINVAL;
- }
-
- /* set the MCK divider for BCLK */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div);
- if (ret < 0)
- return ret;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* set the BCLK divider for DACLRC */
- ret = snd_soc_dai_set_clkdiv(cpu_dai,
- ATMEL_SSC_TCMR_PERIOD, period);
- } else {
- /* set the BCLK divider for ADCLRC */
- ret = snd_soc_dai_set_clkdiv(cpu_dai,
- ATMEL_SSC_RCMR_PERIOD, period);
- }
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
@@ -190,9 +89,7 @@ static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
}
static struct snd_soc_ops at91sam9g20ek_ops = {
- .startup = at91sam9g20ek_startup,
.hw_params = at91sam9g20ek_hw_params,
- .shutdown = at91sam9g20ek_shutdown,
};
static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
@@ -241,10 +138,20 @@ static const struct snd_soc_dapm_route intercon[] = {
*/
static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
{
+ struct snd_soc_dai *codec_dai = &codec->dai[0];
+ int ret;
+
printk(KERN_DEBUG
"at91sam9g20ek_wm8731 "
": at91sam9g20ek_wm8731_init() called\n");
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+ MCLK_RATE, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret);
+ return ret;
+ }
+
/* Add specific widgets */
snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
@@ -255,8 +162,13 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "RLINEIN");
snd_soc_dapm_nc_pin(codec, "LLINEIN");
- /* always connected */
+#ifdef ENABLE_MIC_INPUT
snd_soc_dapm_enable_pin(codec, "Int Mic");
+#else
+ snd_soc_dapm_nc_pin(codec, "Int Mic");
+#endif
+
+ /* always connected */
snd_soc_dapm_enable_pin(codec, "Ext Spk");
snd_soc_dapm_sync(codec);
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 479d7bd..a521aa9 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -1,8 +1,8 @@
/*
* Au12x0/Au1550 PSC ALSA ASoC audio support.
*
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- * Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <manuel.lauss@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
@@ -19,6 +19,7 @@
#include <linux/module.h>
#include <linux/device.h>
#include <linux/delay.h>
+#include <linux/mutex.h>
#include <linux/suspend.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -29,6 +30,9 @@
#include "psc.h"
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
#define AC97_DIR \
(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
@@ -45,6 +49,9 @@
#define AC97PCR_CLRFIFO(stype) \
((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+#define AC97STAT_BUSY(stype) \
+ ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+
/* instance data. There can be only one, MacLeod!!!! */
static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned short data, tmo;
+ unsigned short data, retry, tmo;
- au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
- tmo = 1000;
- while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
- udelay(2);
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&pscdata->lock);
+
+ au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
+ AC97_CDC(pscdata));
+ au_sync();
+
+ tmo = 2000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+ && --tmo)
+ udelay(2);
- if (!tmo)
- data = 0xffff;
- else
data = au_readl(AC97_CDC(pscdata)) & 0xffff;
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+
+ mutex_unlock(&pscdata->lock);
+ } while (--retry && !tmo);
- return data;
+ return retry ? data : 0xffff;
}
/* AC97 controller writes to codec register */
@@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned int tmo;
+ unsigned int tmo, retry;
- au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
au_sync();
- tmo = 1000;
- while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+
+ retry = AC97_RW_RETRIES;
+ do {
+ mutex_lock(&pscdata->lock);
+
+ au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
+ AC97_CDC(pscdata));
au_sync();
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ tmo = 2000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+ && --tmo)
+ udelay(2);
+
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+
+ mutex_unlock(&pscdata->lock);
+ } while (--retry && !tmo);
}
/* AC97 controller asserts a warm reset */
@@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
au_sync();
/* wait for PSC to indicate it's ready */
- i = 100000;
+ i = 1000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
- au_sync();
+ msleep(1);
if (i == 0) {
printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
@@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
au_sync();
/* wait for AC97 core to become ready */
- i = 100000;
+ i = 1000;
while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
- au_sync();
+ msleep(1);
if (i == 0)
printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
}
@@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
{
/* FIXME */
struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
- unsigned long r, stat;
+ unsigned long r, ro, stat;
int chans, stype = SUBSTREAM_TYPE(substream);
chans = params_channels(params);
- r = au_readl(AC97_CFG(pscdata));
+ r = ro = au_readl(AC97_CFG(pscdata));
stat = au_readl(AC97_STAT(pscdata));
/* already active? */
@@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
(pscdata->rate != params_rate(params)))
return -EINVAL;
} else {
- /* disable AC97 device controller first */
- au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
- au_sync();
/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
r &= ~PSC_AC97CFG_LEN_MASK;
@@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
r |= PSC_AC97CFG_RXSLOT_ENA(4);
}
- /* finally enable the AC97 controller again */
+ /* do we need to poke the hardware? */
+ if (!(r ^ ro))
+ goto out;
+
+ /* ac97 engine is about to be disabled */
+ mutex_lock(&pscdata->lock);
+
+ /* disable AC97 device controller first... */
+ au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ au_sync();
+
+ /* ...wait for it... */
+ while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)
+ asm volatile ("nop");
+
+ /* ...write config... */
+ au_writel(r, AC97_CFG(pscdata));
+ au_sync();
+
+ /* ...enable the AC97 controller again... */
au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
au_sync();
+ /* ...and wait for ready bit */
+ while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR))
+ asm volatile ("nop");
+
+ mutex_unlock(&pscdata->lock);
+
pscdata->cfg = r;
pscdata->rate = params_rate(params);
}
+out:
return 0;
}
@@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
+ au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ au_sync();
au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
au_sync();
break;
@@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_SUSPEND:
au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
au_sync();
+
+ while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
+ asm volatile ("nop");
+
+ au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ au_sync();
+
break;
default:
ret = -EINVAL;
@@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
if (!au1xpsc_ac97_workdata)
return -ENOMEM;
+ mutex_init(&au1xpsc_ac97_workdata->lock);
+
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!r) {
ret = -ENODEV;
@@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
goto out1;
/* configuration: max dma trigger threshold, enable ac97 */
- au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
- PSC_AC97CFG_TT_FIFO8 |
- PSC_AC97CFG_DE_ENABLE;
+ au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
+ PSC_AC97CFG_TT_FIFO8 |
+ PSC_AC97CFG_DE_ENABLE;
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
@@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 8fdb1a04..3f474e8 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -29,6 +29,7 @@ struct au1xpsc_audio_data {
unsigned long pm[2];
struct resource *ioarea;
+ struct mutex lock;
};
#define PCM_TX 0
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 811596f..ac927ff 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,6 +7,15 @@ config SND_BF5XX_I2S
mode (supports single stereo In/Out).
You will also need to select the audio interfaces to support below.
+config SND_BF5XX_TDM
+ tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+ depends on (BLACKFIN && SND_SOC)
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in TDM
+ mode.
+ You will also need to select the audio interfaces to support below.
+
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
depends on SND_BF5XX_I2S
@@ -69,12 +78,24 @@ config SND_BF5XX_SOC_I2S
tristate
select SND_BF5XX_SOC_SPORT
+config SND_BF5XX_SOC_TDM
+ tristate
+ select SND_BF5XX_SOC_SPORT
+
config SND_BF5XX_SOC_AC97
tristate
select AC97_BUS
select SND_SOC_AC97_BUS
select SND_BF5XX_SOC_SPORT
+config SND_BF5XX_SOC_AD1836
+ tristate "SoC AD1836 Audio support for BF5xx"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1836
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
config SND_BF5XX_SOC_AD1980
tristate "SoC AD1980/1 Audio support for BF5xx"
depends on SND_BF5XX_AC97
@@ -83,9 +104,17 @@ config SND_BF5XX_SOC_AD1980
help
Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+config SND_BF5XX_SOC_AD1938
+ tristate "SoC AD1938 Audio support for Blackfin"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1938
+ help
+ Say Y if you want to add support for AD1938 codec on Blackfin.
+
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
- depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
+ depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
range 0 3 if BF54x
range 0 1 if !BF54x
default 0
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 97bb37a..87e3042 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -1,21 +1,29 @@
# Blackfin Platform Support
snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o
snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o
+snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o
snd-soc-bf5xx-sport-objs := bf5xx-sport.o
snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
+snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o
obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o
obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
# Blackfin Machine Support
+snd-ad1836-objs := bf5xx-ad1836.o
snd-ad1980-objs := bf5xx-ad1980.o
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
+snd-ad1938-objs := bf5xx-ad1938.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index b1ed423..2758b90 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -277,28 +277,24 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
if (!dai->active)
return 0;
- ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+ ret = sport_set_multichannel(sport, 16, 0x1F, 1);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
+ ret = sport_config_rx(sport, IRFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
+ ret = sport_config_tx(sport, ITFS, 0xF, 0, (16*16-1));
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- if (dai->capture.active)
- sport_rx_start(sport);
- if (dai->playback.active)
- sport_tx_start(sport);
return 0;
}
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
new file mode 100644
index 0000000..cd361e3
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -0,0 +1,128 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ad1836.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Aug 4 2009
+ * Description: Board driver for ad1836 sound chip
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad1836.h"
+#include "bf5xx-sport.h"
+
+#include "bf5xx-tdm-pcm.h"
+#include "bf5xx-tdm.h"
+
+static struct snd_soc_card bf5xx_ad1836;
+
+static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ int ret = 0;
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops bf5xx_ad1836_ops = {
+ .startup = bf5xx_ad1836_startup,
+ .hw_params = bf5xx_ad1836_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad1836_dai = {
+ .name = "ad1836",
+ .stream_name = "AD1836",
+ .cpu_dai = &bf5xx_tdm_dai,
+ .codec_dai = &ad1836_dai,
+ .ops = &bf5xx_ad1836_ops,
+};
+
+static struct snd_soc_card bf5xx_ad1836 = {
+ .name = "bf5xx_ad1836",
+ .platform = &bf5xx_tdm_soc_platform,
+ .dai_link = &bf5xx_ad1836_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad1836_snd_devdata = {
+ .card = &bf5xx_ad1836,
+ .codec_dev = &soc_codec_dev_ad1836,
+};
+
+static struct platform_device *bfxx_ad1836_snd_device;
+
+static int __init bf5xx_ad1836_init(void)
+{
+ int ret;
+
+ bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bfxx_ad1836_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata);
+ bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev;
+ ret = platform_device_add(bfxx_ad1836_snd_device);
+
+ if (ret)
+ platform_device_put(bfxx_ad1836_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_ad1836_exit(void)
+{
+ platform_device_unregister(bfxx_ad1836_snd_device);
+}
+
+module_init(bf5xx_ad1836_init);
+module_exit(bf5xx_ad1836_exit);
+
+/* Module information */
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("ALSA SoC AD1836 board driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c
new file mode 100644
index 0000000..08269e9
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad1938.c
@@ -0,0 +1,142 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ad1938.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Thur June 4 2009
+ * Description: Board driver for ad1938 sound chip
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad1938.h"
+#include "bf5xx-sport.h"
+
+#include "bf5xx-tdm-pcm.h"
+#include "bf5xx-tdm.h"
+
+static struct snd_soc_card bf5xx_ad1938;
+
+static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ int ret = 0;
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set codec DAI slots, 8 channels, all channels are enabled */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops bf5xx_ad1938_ops = {
+ .startup = bf5xx_ad1938_startup,
+ .hw_params = bf5xx_ad1938_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad1938_dai = {
+ .name = "ad1938",
+ .stream_name = "AD1938",
+ .cpu_dai = &bf5xx_tdm_dai,
+ .codec_dai = &ad1938_dai,
+ .ops = &bf5xx_ad1938_ops,
+};
+
+static struct snd_soc_card bf5xx_ad1938 = {
+ .name = "bf5xx_ad1938",
+ .platform = &bf5xx_tdm_soc_platform,
+ .dai_link = &bf5xx_ad1938_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad1938_snd_devdata = {
+ .card = &bf5xx_ad1938,
+ .codec_dev = &soc_codec_dev_ad1938,
+};
+
+static struct platform_device *bfxx_ad1938_snd_device;
+
+static int __init bf5xx_ad1938_init(void)
+{
+ int ret;
+
+ bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bfxx_ad1938_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata);
+ bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev;
+ ret = platform_device_add(bfxx_ad1938_snd_device);
+
+ if (ret)
+ platform_device_put(bfxx_ad1938_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_ad1938_exit(void)
+{
+ platform_device_unregister(bfxx_ad1938_snd_device);
+}
+
+module_init(bf5xx_ad1938_init);
+module_exit(bf5xx_ad1938_exit);
+
+/* Module information */
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("ALSA SoC AD1938 board driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index edfbdc0..9825b71 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
.codec_dev = &soc_codec_dev_ad73311,
};
-static struct platform_device *bf52x_ad73311_snd_device;
+static struct platform_device *bf5xx_ad73311_snd_device;
static int __init bf5xx_ad73311_init(void)
{
int ret;
pr_debug("%s enter\n", __func__);
- bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
- if (!bf52x_ad73311_snd_device)
+ bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bf5xx_ad73311_snd_device)
return -ENOMEM;
- platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
- bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
- ret = platform_device_add(bf52x_ad73311_snd_device);
+ platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
+ bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev;
+ ret = platform_device_add(bf5xx_ad73311_snd_device);
if (ret)
- platform_device_put(bf52x_ad73311_snd_device);
+ platform_device_put(bf5xx_ad73311_snd_device);
return ret;
}
@@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void)
static void __exit bf5xx_ad73311_exit(void)
{
pr_debug("%s enter\n", __func__);
- platform_device_unregister(bf52x_ad73311_snd_device);
+ platform_device_unregister(bf5xx_ad73311_snd_device);
}
module_init(bf5xx_ad73311_init);
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index af06904..876abad 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -259,22 +259,18 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
if (!dai->active)
return 0;
- ret = sport_config_rx(sport_handle, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+ ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- ret = sport_config_tx(sport_handle, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+ ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
if (ret) {
pr_err("SPORT is busy!\n");
return -EBUSY;
}
- if (dai->capture.active)
- sport_rx_start(sport);
- if (dai->playback.active)
- sport_tx_start(sport);
return 0;
}
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index bc0cdde..3a00fa4 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
.codec_data = &bf5xx_ssm2602_setup,
};
-static struct platform_device *bf52x_ssm2602_snd_device;
+static struct platform_device *bf5xx_ssm2602_snd_device;
static int __init bf5xx_ssm2602_init(void)
{
int ret;
pr_debug("%s enter\n", __func__);
- bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
- if (!bf52x_ssm2602_snd_device)
+ bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bf5xx_ssm2602_snd_device)
return -ENOMEM;
- platform_set_drvdata(bf52x_ssm2602_snd_device,
+ platform_set_drvdata(bf5xx_ssm2602_snd_device,
&bf5xx_ssm2602_snd_devdata);
- bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev;
- ret = platform_device_add(bf52x_ssm2602_snd_device);
+ bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev;
+ ret = platform_device_add(bf5xx_ssm2602_snd_device);
if (ret)
- platform_device_put(bf52x_ssm2602_snd_device);
+ platform_device_put(bf5xx_ssm2602_snd_device);
return ret;
}
@@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void)
static void __exit bf5xx_ssm2602_exit(void)
{
pr_debug("%s enter\n", __func__);
- platform_device_unregister(bf52x_ssm2602_snd_device);
+ platform_device_unregister(bf5xx_ssm2602_snd_device);
}
module_init(bf5xx_ssm2602_init);
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
new file mode 100644
index 0000000..ccb5e82
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -0,0 +1,330 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-tdm-pcm.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Tue June 06 2009
+ * Description: DMA driver for tdm codec
+ *
+ * Modified:
+ * Copyright 2009 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+
+#include "bf5xx-tdm-pcm.h"
+#include "bf5xx-tdm.h"
+#include "bf5xx-sport.h"
+
+#define PCM_BUFFER_MAX 0x10000
+#define FRAGMENT_SIZE_MIN (4*1024)
+#define FRAGMENTS_MIN 2
+#define FRAGMENTS_MAX 32
+
+static void bf5xx_dma_irq(void *data)
+{
+ struct snd_pcm_substream *pcm = data;
+ snd_pcm_period_elapsed(pcm);
+}
+
+static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .channels_min = 2,
+ .channels_max = 8,
+ .buffer_bytes_max = PCM_BUFFER_MAX,
+ .period_bytes_min = FRAGMENT_SIZE_MIN,
+ .period_bytes_max = PCM_BUFFER_MAX/2,
+ .periods_min = FRAGMENTS_MIN,
+ .periods_max = FRAGMENTS_MAX,
+};
+
+static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max;
+ snd_pcm_lib_malloc_pages(substream, size * 4);
+
+ return 0;
+}
+
+static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size);
+
+ fragsize_bytes /= runtime->channels;
+ /* inflate the fragsize to match the dma width of SPORT */
+ fragsize_bytes *= 8;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_tx_dma(sport, runtime->dma_area,
+ runtime->periods, fragsize_bytes);
+ } else {
+ sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_rx_dma(sport, runtime->dma_area,
+ runtime->periods, fragsize_bytes);
+ }
+
+ return 0;
+}
+
+static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_tx_start(sport);
+ else
+ sport_rx_start(sport);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_tx_stop(sport);
+ else
+ sport_rx_stop(sport);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ unsigned int diff;
+ snd_pcm_uframes_t frames;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ diff = sport_curr_offset_tx(sport);
+ frames = diff / (8*4); /* 32 bytes per frame */
+ } else {
+ diff = sport_curr_offset_rx(sport);
+ frames = diff / (8*4);
+ }
+ return frames;
+}
+
+static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ if (sport_handle != NULL)
+ runtime->private_data = sport_handle;
+ else {
+ pr_err("sport_handle is NULL\n");
+ ret = -ENODEV;
+ }
+out:
+ return ret;
+}
+
+static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count)
+{
+ unsigned int *src;
+ unsigned int *dst;
+ int i;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ src = buf;
+ dst = (unsigned int *)substream->runtime->dma_area;
+
+ dst += pos * 8;
+ while (count--) {
+ for (i = 0; i < substream->runtime->channels; i++)
+ *(dst + i) = *src++;
+ dst += 8;
+ }
+ } else {
+ src = (unsigned int *)substream->runtime->dma_area;
+ dst = buf;
+
+ src += pos * 8;
+ while (count--) {
+ for (i = 0; i < substream->runtime->channels; i++)
+ *dst++ = *(src+i);
+ src += 8;
+ }
+ }
+
+ return 0;
+}
+
+static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count)
+{
+ unsigned char *buf = substream->runtime->dma_area;
+ buf += pos * 8 * 4;
+ memset(buf, '\0', count * 8 * 4);
+
+ return 0;
+}
+
+
+struct snd_pcm_ops bf5xx_pcm_tdm_ops = {
+ .open = bf5xx_pcm_open,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = bf5xx_pcm_hw_params,
+ .hw_free = bf5xx_pcm_hw_free,
+ .prepare = bf5xx_pcm_prepare,
+ .trigger = bf5xx_pcm_trigger,
+ .pointer = bf5xx_pcm_pointer,
+ .copy = bf5xx_pcm_copy,
+ .silence = bf5xx_pcm_silence,
+};
+
+static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size * 4,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area) {
+ pr_err("Failed to allocate dma memory \
+ Please increase uncached DMA memory region\n");
+ return -ENOMEM;
+ }
+ buf->bytes = size;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_handle->tx_buf = buf->area;
+ else
+ sport_handle->rx_buf = buf->area;
+
+ return 0;
+}
+
+static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_coherent(NULL, buf->bytes, buf->area, 0);
+ buf->area = NULL;
+ }
+ if (sport_handle)
+ sport_done(sport_handle);
+}
+
+static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &bf5xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (dai->playback.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+out:
+ return ret;
+}
+
+struct snd_soc_platform bf5xx_tdm_soc_platform = {
+ .name = "bf5xx-audio",
+ .pcm_ops = &bf5xx_pcm_tdm_ops,
+ .pcm_new = bf5xx_pcm_tdm_new,
+ .pcm_free = bf5xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(bf5xx_tdm_soc_platform);
+
+static int __init bfin_pcm_tdm_init(void)
+{
+ return snd_soc_register_platform(&bf5xx_tdm_soc_platform);
+}
+module_init(bfin_pcm_tdm_init);
+
+static void __exit bfin_pcm_tdm_exit(void)
+{
+ snd_soc_unregister_platform(&bf5xx_tdm_soc_platform);
+}
+module_exit(bfin_pcm_tdm_exit);
+
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h
new file mode 100644
index 0000000..ddc5047
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h
@@ -0,0 +1,21 @@
+/*
+ * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin
+ *
+ * Copyright 2009 Analog Device Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_TDM_PCM_H
+#define _BF5XX_TDM_PCM_H
+
+struct bf5xx_pcm_dma_params {
+ char *name; /* stream identifier */
+};
+
+/* platform data */
+extern struct snd_soc_platform bf5xx_tdm_soc_platform;
+
+#endif
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
new file mode 100644
index 0000000..3096bad
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -0,0 +1,343 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-tdm.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Thurs June 04 2009
+ * Description: Blackfin I2S(TDM) CPU DAI driver
+ * Even though TDM mode can be as part of I2S DAI, but there
+ * are so much difference in configuration and data flow,
+ * it's very ugly to integrate I2S and TDM into a module
+ *
+ * Modified:
+ * Copyright 2009 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/irq.h>
+#include <asm/portmux.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+
+#include "bf5xx-sport.h"
+#include "bf5xx-tdm.h"
+
+struct bf5xx_tdm_port {
+ u16 tcr1;
+ u16 rcr1;
+ u16 tcr2;
+ u16 rcr2;
+ int configured;
+};
+
+static struct bf5xx_tdm_port bf5xx_tdm;
+static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
+
+static struct sport_param sport_params[2] = {
+ {
+ .dma_rx_chan = CH_SPORT0_RX,
+ .dma_tx_chan = CH_SPORT0_TX,
+ .err_irq = IRQ_SPORT0_ERROR,
+ .regs = (struct sport_register *)SPORT0_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT1_RX,
+ .dma_tx_chan = CH_SPORT1_TX,
+ .err_irq = IRQ_SPORT1_ERROR,
+ .regs = (struct sport_register *)SPORT1_TCR1,
+ }
+};
+
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS. Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+ {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+ P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
+
+static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ int ret = 0;
+
+ /* interface format:support TDM,slave mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ break;
+ default:
+ printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ret = -EINVAL;
+ break;
+ default:
+ printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int ret = 0;
+
+ bf5xx_tdm.tcr2 &= ~0x1f;
+ bf5xx_tdm.rcr2 &= ~0x1f;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bf5xx_tdm.tcr2 |= 31;
+ bf5xx_tdm.rcr2 |= 31;
+ sport_handle->wdsize = 4;
+ break;
+ /* at present, we only support 32bit transfer */
+ default:
+ pr_err("not supported PCM format yet\n");
+ return -EINVAL;
+ break;
+ }
+
+ if (!bf5xx_tdm.configured) {
+ /*
+ * TX and RX are not independent,they are enabled at the
+ * same time, even if only one side is running. So, we
+ * need to configure both of them at the time when the first
+ * stream is opened.
+ *
+ * CPU DAI:slave mode.
+ */
+ ret = sport_config_rx(sport_handle, bf5xx_tdm.rcr1,
+ bf5xx_tdm.rcr2, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ ret = sport_config_tx(sport_handle, bf5xx_tdm.tcr1,
+ bf5xx_tdm.tcr2, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ bf5xx_tdm.configured = 1;
+ }
+
+ return 0;
+}
+
+static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ /* No active stream, SPORT is allowed to be configured again. */
+ if (!dai->active)
+ bf5xx_tdm.configured = 0;
+}
+
+#ifdef CONFIG_PM
+static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
+{
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ if (!dai->active)
+ return 0;
+ if (dai->capture.active)
+ sport_rx_stop(sport);
+ if (dai->playback.active)
+ sport_tx_stop(sport);
+ return 0;
+}
+
+static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
+{
+ int ret;
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ if (!dai->active)
+ return 0;
+
+ ret = sport_set_multichannel(sport, 8, 0xFF, 1);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ }
+
+ ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ }
+
+ ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ }
+
+ return 0;
+}
+
+#else
+#define bf5xx_tdm_suspend NULL
+#define bf5xx_tdm_resume NULL
+#endif
+
+static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = {
+ .hw_params = bf5xx_tdm_hw_params,
+ .set_fmt = bf5xx_tdm_set_dai_fmt,
+ .shutdown = bf5xx_tdm_shutdown,
+};
+
+struct snd_soc_dai bf5xx_tdm_dai = {
+ .name = "bf5xx-tdm",
+ .id = 0,
+ .suspend = bf5xx_tdm_suspend,
+ .resume = bf5xx_tdm_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .ops = &bf5xx_tdm_dai_ops,
+};
+EXPORT_SYMBOL_GPL(bf5xx_tdm_dai);
+
+static int __devinit bfin_tdm_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
+ pr_err("Requesting Peripherals failed\n");
+ return -EFAULT;
+ }
+
+ /* request DMA for SPORT */
+ sport_handle = sport_init(&sport_params[sport_num], 4, \
+ 8 * sizeof(u32), NULL);
+ if (!sport_handle) {
+ peripheral_free_list(&sport_req[sport_num][0]);
+ return -ENODEV;
+ }
+
+ /* SPORT works in TDM mode */
+ ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ goto sport_config_err;
+ }
+
+ ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ goto sport_config_err;
+ }
+
+ ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ ret = -EBUSY;
+ goto sport_config_err;
+ }
+
+ ret = snd_soc_register_dai(&bf5xx_tdm_dai);
+ if (ret) {
+ pr_err("Failed to register DAI: %d\n", ret);
+ goto sport_config_err;
+ }
+ return 0;
+
+sport_config_err:
+ peripheral_free_list(&sport_req[sport_num][0]);
+ return ret;
+}
+
+static int __devexit bfin_tdm_remove(struct platform_device *pdev)
+{
+ peripheral_free_list(&sport_req[sport_num][0]);
+ snd_soc_unregister_dai(&bf5xx_tdm_dai);
+
+ return 0;
+}
+
+static struct platform_driver bfin_tdm_driver = {
+ .probe = bfin_tdm_probe,
+ .remove = __devexit_p(bfin_tdm_remove),
+ .driver = {
+ .name = "bfin-tdm",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init bfin_tdm_init(void)
+{
+ return platform_driver_register(&bfin_tdm_driver);
+}
+module_init(bfin_tdm_init);
+
+static void __exit bfin_tdm_exit(void)
+{
+ platform_driver_unregister(&bfin_tdm_driver);
+}
+module_exit(bfin_tdm_exit);
+
+/* Module information */
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("TDM driver for ADI Blackfin");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h
new file mode 100644
index 0000000..618ec3d
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-tdm.h
@@ -0,0 +1,14 @@
+/*
+ * sound/soc/blackfin/bf5xx-tdm.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_TDM_H
+#define _BF5XX_TDM_H
+
+extern struct snd_soc_dai bf5xx_tdm_dai;
+
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bbc97fd..0edca93 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -12,11 +12,15 @@ config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
select SND_SOC_L3
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+ select SND_SOC_AD1836 if SPI_MASTER
+ select SND_SOC_AD1938 if SPI_MASTER
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311 if I2C
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
+ select SND_SOC_AK4642 if I2C
select SND_SOC_CS4270 if I2C
+ select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
@@ -30,18 +34,23 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8523 if I2C
select SND_SOC_WM8580 if I2C
select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
select SND_SOC_WM8940 if I2C
select SND_SOC_WM8960 if I2C
+ select SND_SOC_WM8961 if I2C
select SND_SOC_WM8971 if I2C
+ select SND_SOC_WM8974 if I2C
select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM8993 if I2C
select SND_SOC_WM9081 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
@@ -57,11 +66,21 @@ config SND_SOC_ALL_CODECS
If unsure select "N".
+config SND_SOC_WM_HUBS
+ tristate
+ default y if SND_SOC_WM8993=y
+ default m if SND_SOC_WM8993=m
config SND_SOC_AC97_CODEC
tristate
select SND_AC97_CODEC
+config SND_SOC_AD1836
+ tristate
+
+config SND_SOC_AD1938
+ tristate
+
config SND_SOC_AD1980
tristate
@@ -74,6 +93,9 @@ config SND_SOC_AK4104
config SND_SOC_AK4535
tristate
+config SND_SOC_AK4642
+ tristate
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
@@ -86,6 +108,9 @@ config SND_SOC_CS4270_VD33_ERRATA
bool
depends on SND_SOC_CS4270
+config SND_SOC_CX20442
+ tristate
+
config SND_SOC_L3
tristate
@@ -129,6 +154,9 @@ config SND_SOC_WM8400
config SND_SOC_WM8510
tristate
+config SND_SOC_WM8523
+ tristate
+
config SND_SOC_WM8580
tristate
@@ -144,6 +172,9 @@ config SND_SOC_WM8750
config SND_SOC_WM8753
tristate
+config SND_SOC_WM8776
+ tristate
+
config SND_SOC_WM8900
tristate
@@ -156,15 +187,24 @@ config SND_SOC_WM8940
config SND_SOC_WM8960
tristate
+config SND_SOC_WM8961
+ tristate
+
config SND_SOC_WM8971
tristate
+config SND_SOC_WM8974
+ tristate
+
config SND_SOC_WM8988
tristate
config SND_SOC_WM8990
tristate
+config SND_SOC_WM8993
+ tristate
+
config SND_SOC_WM9081
tristate
@@ -176,3 +216,7 @@ config SND_SOC_WM9712
config SND_SOC_WM9713
tristate
+
+# Amp
+config SND_SOC_MAX9877
+ tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 8b75305..fb4af28 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,9 +1,13 @@
snd-soc-ac97-objs := ac97.o
+snd-soc-ad1836-objs := ad1836.o
+snd-soc-ad1938-objs := ad1938.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4642-objs := ak4642.o
snd-soc-cs4270-objs := cs4270.o
+snd-soc-cx20442-objs := cx20442.o
snd-soc-l3-objs := l3.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-spdif-objs := spdif_transciever.o
@@ -18,29 +22,42 @@ snd-soc-uda1380-objs := uda1380.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
snd-soc-wm8510-objs := wm8510.o
+snd-soc-wm8523-objs := wm8523.o
snd-soc-wm8580-objs := wm8580.o
snd-soc-wm8728-objs := wm8728.o
snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8776-objs := wm8776.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
snd-soc-wm8940-objs := wm8940.o
snd-soc-wm8960-objs := wm8960.o
+snd-soc-wm8961-objs := wm8961.o
snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8974-objs := wm8974.o
snd-soc-wm8988-objs := wm8988.o
snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm8993-objs := wm8993.o
snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
+snd-soc-wm-hubs-objs := wm_hubs.o
+
+# Amp
+snd-soc-max9877-objs := max9877.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
+obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
@@ -55,19 +72,28 @@ obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
+obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o
obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o
obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o
obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o
obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o
obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o
obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
+obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o
+
+# Amp
+obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
new file mode 100644
index 0000000..3612bb9
--- /dev/null
+++ b/sound/soc/codecs/ad1836.c
@@ -0,0 +1,446 @@
+/*
+ * File: sound/soc/codecs/ad1836.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Aug 04 2009
+ * Description: Driver for AD1836 sound chip
+ *
+ * Modified:
+ * Copyright 2009 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-dapm.h>
+#include <linux/spi/spi.h>
+#include "ad1836.h"
+
+/* codec private data */
+struct ad1836_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[AD1836_NUM_REGS];
+};
+
+static struct snd_soc_codec *ad1836_codec;
+struct snd_soc_codec_device soc_codec_dev_ad1836;
+static int ad1836_register(struct ad1836_priv *ad1836);
+static void ad1836_unregister(struct ad1836_priv *ad1836);
+
+/*
+ * AD1836 volume/mute/de-emphasis etc. controls
+ */
+static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"};
+
+static const struct soc_enum ad1836_deemp_enum =
+ SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
+
+static const struct snd_kcontrol_new ad1836_snd_controls[] = {
+ /* DAC volume control */
+ SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
+ AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
+ SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
+ AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
+ SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
+ AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
+
+ /* ADC switch control */
+ SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
+ AD1836_ADCR1_MUTE, 1, 1),
+ SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
+ AD1836_ADCR2_MUTE, 1, 1),
+
+ /* DAC switch control */
+ SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
+ AD1836_DACR1_MUTE, 1, 1),
+ SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
+ AD1836_DACR2_MUTE, 1, 1),
+ SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
+ AD1836_DACR3_MUTE, 1, 1),
+
+ /* ADC high-pass filter */
+ SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
+ AD1836_ADC_HIGHPASS_FILTER, 1, 0),
+
+ /* DAC de-emphasis */
+ SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
+};
+
+static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
+ AD1836_DAC_POWERDOWN, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
+ AD1836_ADC_POWERDOWN, 1, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+ SND_SOC_DAPM_INPUT("ADC1IN"),
+ SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "DAC", NULL, "ADC_PWR" },
+ { "ADC", NULL, "ADC_PWR" },
+ { "DAC1OUT", "DAC1 Switch", "DAC" },
+ { "DAC2OUT", "DAC2 Switch", "DAC" },
+ { "DAC3OUT", "DAC3 Switch", "DAC" },
+ { "ADC", "ADC1 Switch", "ADC1IN" },
+ { "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+/*
+ * DAI ops entries
+ */
+
+static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ /* at present, we support adc aux mode to interface with
+ * blackfin sport tdm mode
+ */
+ case SND_SOC_DAIFMT_DSP_A:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ /* ALCLK,ABCLK are both output, AD1836 can only be master */
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ad1836_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int word_len = 0;
+
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ word_len = 3;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ word_len = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S32_LE:
+ word_len = 0;
+ break;
+ }
+
+ snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
+ AD1836_DAC_WORD_LEN_MASK, word_len);
+
+ snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
+ AD1836_ADC_WORD_LEN_MASK, word_len);
+
+ return 0;
+}
+
+
+/*
+ * interface to read/write ad1836 register
+ */
+#define AD1836_SPI_REG_SHFT 12
+#define AD1836_SPI_READ (1 << 11)
+#define AD1836_SPI_VAL_MSK 0x3FF
+
+/*
+ * write to the ad1836 register space
+ */
+
+static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *reg_cache = codec->reg_cache;
+ int ret = 0;
+
+ if (value != reg_cache[reg]) {
+ unsigned short buf;
+ struct spi_transfer t = {
+ .tx_buf = &buf,
+ .len = 2,
+ };
+ struct spi_message m;
+
+ buf = (reg << AD1836_SPI_REG_SHFT) |
+ (value & AD1836_SPI_VAL_MSK);
+ spi_message_init(&m);
+ spi_message_add_tail(&t, &m);
+ ret = spi_sync(codec->control_data, &m);
+ if (ret == 0)
+ reg_cache[reg] = value;
+ }
+
+ return ret;
+}
+
+/*
+ * read from the ad1836 register space cache
+ */
+static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *reg_cache = codec->reg_cache;
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ return reg_cache[reg];
+}
+
+static int __devinit ad1836_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_codec *codec;
+ struct ad1836_priv *ad1836;
+
+ ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL);
+ if (ad1836 == NULL)
+ return -ENOMEM;
+
+ codec = &ad1836->codec;
+ codec->control_data = spi;
+ codec->dev = &spi->dev;
+
+ dev_set_drvdata(&spi->dev, ad1836);
+
+ return ad1836_register(ad1836);
+}
+
+static int __devexit ad1836_spi_remove(struct spi_device *spi)
+{
+ struct ad1836_priv *ad1836 = dev_get_drvdata(&spi->dev);
+
+ ad1836_unregister(ad1836);
+ return 0;
+}
+
+static struct spi_driver ad1836_spi_driver = {
+ .driver = {
+ .name = "ad1836-spi",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = ad1836_spi_probe,
+ .remove = __devexit_p(ad1836_spi_remove),
+};
+
+static struct snd_soc_dai_ops ad1836_dai_ops = {
+ .hw_params = ad1836_hw_params,
+ .set_fmt = ad1836_set_dai_fmt,
+};
+
+/* codec DAI instance */
+struct snd_soc_dai ad1836_dai = {
+ .name = "AD1836",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 4,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .ops = &ad1836_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ad1836_dai);
+
+static int ad1836_register(struct ad1836_priv *ad1836)
+{
+ int ret;
+ struct snd_soc_codec *codec = &ad1836->codec;
+
+ if (ad1836_codec) {
+ dev_err(codec->dev, "Another ad1836 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ codec->private_data = ad1836;
+ codec->reg_cache = ad1836->reg_cache;
+ codec->reg_cache_size = AD1836_NUM_REGS;
+ codec->name = "AD1836";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ad1836_dai;
+ codec->num_dai = 1;
+ codec->write = ad1836_write_reg;
+ codec->read = ad1836_read_reg_cache;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ad1836_dai.dev = codec->dev;
+ ad1836_codec = codec;
+
+ /* default setting for ad1836 */
+ /* de-emphasis: 48kHz, power-on dac */
+ codec->write(codec, AD1836_DAC_CTRL1, 0x300);
+ /* unmute dac channels */
+ codec->write(codec, AD1836_DAC_CTRL2, 0x0);
+ /* high-pass filter enable, power-on adc */
+ codec->write(codec, AD1836_ADC_CTRL1, 0x100);
+ /* unmute adc channles, adc aux mode */
+ codec->write(codec, AD1836_ADC_CTRL2, 0x180);
+ /* left/right diff:PGA/MUX */
+ codec->write(codec, AD1836_ADC_CTRL3, 0x3A);
+ /* volume */
+ codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF);
+ codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF);
+ codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF);
+ codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF);
+ codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF);
+ codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF);
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ kfree(ad1836);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&ad1836_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ kfree(ad1836);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void ad1836_unregister(struct ad1836_priv *ad1836)
+{
+ snd_soc_unregister_dai(&ad1836_dai);
+ snd_soc_unregister_codec(&ad1836->codec);
+ kfree(ad1836);
+ ad1836_codec = NULL;
+}
+
+static int ad1836_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (ad1836_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ad1836_codec;
+ codec = ad1836_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, ad1836_snd_controls,
+ ARRAY_SIZE(ad1836_snd_controls));
+ snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets,
+ ARRAY_SIZE(ad1836_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int ad1836_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad1836 = {
+ .probe = ad1836_probe,
+ .remove = ad1836_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836);
+
+static int __init ad1836_init(void)
+{
+ int ret;
+
+ ret = spi_register_driver(&ad1836_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(ad1836_init);
+
+static void __exit ad1836_exit(void)
+{
+ spi_unregister_driver(&ad1836_spi_driver);
+}
+module_exit(ad1836_exit);
+
+MODULE_DESCRIPTION("ASoC ad1836 driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
new file mode 100644
index 0000000..7660ee6
--- /dev/null
+++ b/sound/soc/codecs/ad1836.h
@@ -0,0 +1,64 @@
+/*
+ * File: sound/soc/codecs/ad1836.h
+ * Based on:
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: Aug 04, 2009
+ * Description: definitions for AD1836 registers
+ *
+ * Modified:
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#ifndef __AD1836_H__
+#define __AD1836_H__
+
+#define AD1836_DAC_CTRL1 0
+#define AD1836_DAC_POWERDOWN 2
+#define AD1836_DAC_SERFMT_MASK 0xE0
+#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5)
+#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5)
+#define AD1836_DAC_WORD_LEN_MASK 0x18
+
+#define AD1836_DAC_CTRL2 1
+#define AD1836_DACL1_MUTE 0
+#define AD1836_DACR1_MUTE 1
+#define AD1836_DACL2_MUTE 2
+#define AD1836_DACR2_MUTE 3
+#define AD1836_DACL3_MUTE 4
+#define AD1836_DACR3_MUTE 5
+
+#define AD1836_DAC_L1_VOL 2
+#define AD1836_DAC_R1_VOL 3
+#define AD1836_DAC_L2_VOL 4
+#define AD1836_DAC_R2_VOL 5
+#define AD1836_DAC_L3_VOL 6
+#define AD1836_DAC_R3_VOL 7
+
+#define AD1836_ADC_CTRL1 12
+#define AD1836_ADC_POWERDOWN 7
+#define AD1836_ADC_HIGHPASS_FILTER 8
+
+#define AD1836_ADC_CTRL2 13
+#define AD1836_ADCL1_MUTE 0
+#define AD1836_ADCR1_MUTE 1
+#define AD1836_ADCL2_MUTE 2
+#define AD1836_ADCR2_MUTE 3
+#define AD1836_ADC_WORD_LEN_MASK 0x30
+#define AD1836_ADC_SERFMT_MASK (7 << 6)
+#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
+#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
+
+#define AD1836_ADC_CTRL3 14
+
+#define AD1836_NUM_REGS 16
+
+extern struct snd_soc_dai ad1836_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad1836;
+#endif
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
new file mode 100644
index 0000000..e62b277
--- /dev/null
+++ b/sound/soc/codecs/ad1938.c
@@ -0,0 +1,682 @@
+/*
+ * File: sound/soc/codecs/ad1938.c
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: June 04 2009
+ * Description: Driver for AD1938 sound chip
+ *
+ * Modified:
+ * Copyright 2009 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-dapm.h>
+#include <linux/spi/spi.h>
+#include "ad1938.h"
+
+/* codec private data */
+struct ad1938_priv {
+ struct snd_soc_codec codec;
+ u8 reg_cache[AD1938_NUM_REGS];
+};
+
+static struct snd_soc_codec *ad1938_codec;
+struct snd_soc_codec_device soc_codec_dev_ad1938;
+static int ad1938_register(struct ad1938_priv *ad1938);
+static void ad1938_unregister(struct ad1938_priv *ad1938);
+
+/*
+ * AD1938 volume/mute/de-emphasis etc. controls
+ */
+static const char *ad1938_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"};
+
+static const struct soc_enum ad1938_deemp_enum =
+ SOC_ENUM_SINGLE(AD1938_DAC_CTRL2, 1, 4, ad1938_deemp);
+
+static const struct snd_kcontrol_new ad1938_snd_controls[] = {
+ /* DAC volume control */
+ SOC_DOUBLE_R("DAC1 Volume", AD1938_DAC_L1_VOL,
+ AD1938_DAC_R1_VOL, 0, 0xFF, 1),
+ SOC_DOUBLE_R("DAC2 Volume", AD1938_DAC_L2_VOL,
+ AD1938_DAC_R2_VOL, 0, 0xFF, 1),
+ SOC_DOUBLE_R("DAC3 Volume", AD1938_DAC_L3_VOL,
+ AD1938_DAC_R3_VOL, 0, 0xFF, 1),
+ SOC_DOUBLE_R("DAC4 Volume", AD1938_DAC_L4_VOL,
+ AD1938_DAC_R4_VOL, 0, 0xFF, 1),
+
+ /* ADC switch control */
+ SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE,
+ AD1938_ADCR1_MUTE, 1, 1),
+ SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE,
+ AD1938_ADCR2_MUTE, 1, 1),
+
+ /* DAC switch control */
+ SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE,
+ AD1938_DACR1_MUTE, 1, 1),
+ SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE,
+ AD1938_DACR2_MUTE, 1, 1),
+ SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE,
+ AD1938_DACR3_MUTE, 1, 1),
+ SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE,
+ AD1938_DACR4_MUTE, 1, 1),
+
+ /* ADC high-pass filter */
+ SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0,
+ AD1938_ADC_HIGHPASS_FILTER, 1, 0),
+
+ /* DAC de-emphasis */
+ SOC_ENUM("Playback Deemphasis", ad1938_deemp_enum),
+};
+
+static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+ SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+ SND_SOC_DAPM_INPUT("ADC1IN"),
+ SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "DAC", NULL, "ADC_PWR" },
+ { "ADC", NULL, "ADC_PWR" },
+ { "DAC1OUT", "DAC1 Switch", "DAC" },
+ { "DAC2OUT", "DAC2 Switch", "DAC" },
+ { "DAC3OUT", "DAC3 Switch", "DAC" },
+ { "DAC4OUT", "DAC4 Switch", "DAC" },
+ { "ADC", "ADC1 Switch", "ADC1IN" },
+ { "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+/*
+ * DAI ops entries
+ */
+
+static int ad1938_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int reg;
+
+ reg = codec->read(codec, AD1938_DAC_CTRL2);
+ reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg &
+ (~AD1938_DAC_MASTER_MUTE);
+ codec->write(codec, AD1938_DAC_CTRL2, reg);
+
+ return 0;
+}
+
+static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd)
+{
+ int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0);
+ reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg |
+ AD1938_PLL_POWERDOWN;
+ codec->write(codec, AD1938_PLL_CLK_CTRL0, reg);
+
+ return 0;
+}
+
+static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int mask, int slots, int width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int dac_reg = codec->read(codec, AD1938_DAC_CTRL1);
+ int adc_reg = codec->read(codec, AD1938_ADC_CTRL2);
+
+ dac_reg &= ~AD1938_DAC_CHAN_MASK;
+ adc_reg &= ~AD1938_ADC_CHAN_MASK;
+
+ switch (slots) {
+ case 2:
+ dac_reg |= AD1938_DAC_2_CHANNELS << AD1938_DAC_CHAN_SHFT;
+ adc_reg |= AD1938_ADC_2_CHANNELS << AD1938_ADC_CHAN_SHFT;
+ break;
+ case 4:
+ dac_reg |= AD1938_DAC_4_CHANNELS << AD1938_DAC_CHAN_SHFT;
+ adc_reg |= AD1938_ADC_4_CHANNELS << AD1938_ADC_CHAN_SHFT;
+ break;
+ case 8:
+ dac_reg |= AD1938_DAC_8_CHANNELS << AD1938_DAC_CHAN_SHFT;
+ adc_reg |= AD1938_ADC_8_CHANNELS << AD1938_ADC_CHAN_SHFT;
+ break;
+ case 16:
+ dac_reg |= AD1938_DAC_16_CHANNELS << AD1938_DAC_CHAN_SHFT;
+ adc_reg |= AD1938_ADC_16_CHANNELS << AD1938_ADC_CHAN_SHFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ codec->write(codec, AD1938_DAC_CTRL1, dac_reg);
+ codec->write(codec, AD1938_ADC_CTRL2, adc_reg);
+
+ return 0;
+}
+
+static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int adc_reg, dac_reg;
+
+ adc_reg = codec->read(codec, AD1938_ADC_CTRL2);
+ dac_reg = codec->read(codec, AD1938_DAC_CTRL1);
+
+ /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S
+ * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A)
+ */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ adc_reg &= ~AD1938_ADC_SERFMT_MASK;
+ adc_reg |= AD1938_ADC_SERFMT_TDM;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ adc_reg &= ~AD1938_ADC_SERFMT_MASK;
+ adc_reg |= AD1938_ADC_SERFMT_AUX;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
+ adc_reg &= ~AD1938_ADC_LEFT_HIGH;
+ adc_reg &= ~AD1938_ADC_BCLK_INV;
+ dac_reg &= ~AD1938_DAC_LEFT_HIGH;
+ dac_reg &= ~AD1938_DAC_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */
+ adc_reg |= AD1938_ADC_LEFT_HIGH;
+ adc_reg &= ~AD1938_ADC_BCLK_INV;
+ dac_reg |= AD1938_DAC_LEFT_HIGH;
+ dac_reg &= ~AD1938_DAC_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */
+ adc_reg &= ~AD1938_ADC_LEFT_HIGH;
+ adc_reg |= AD1938_ADC_BCLK_INV;
+ dac_reg &= ~AD1938_DAC_LEFT_HIGH;
+ dac_reg |= AD1938_DAC_BCLK_INV;
+ break;
+
+ case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */
+ adc_reg |= AD1938_ADC_LEFT_HIGH;
+ adc_reg |= AD1938_ADC_BCLK_INV;
+ dac_reg |= AD1938_DAC_LEFT_HIGH;
+ dac_reg |= AD1938_DAC_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
+ adc_reg |= AD1938_ADC_LCR_MASTER;
+ adc_reg |= AD1938_ADC_BCLK_MASTER;
+ dac_reg |= AD1938_DAC_LCR_MASTER;
+ dac_reg |= AD1938_DAC_BCLK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */
+ adc_reg |= AD1938_ADC_LCR_MASTER;
+ adc_reg &= ~AD1938_ADC_BCLK_MASTER;
+ dac_reg |= AD1938_DAC_LCR_MASTER;
+ dac_reg &= ~AD1938_DAC_BCLK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
+ adc_reg &= ~AD1938_ADC_LCR_MASTER;
+ adc_reg |= AD1938_ADC_BCLK_MASTER;
+ dac_reg &= ~AD1938_DAC_LCR_MASTER;
+ dac_reg |= AD1938_DAC_BCLK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */
+ adc_reg &= ~AD1938_ADC_LCR_MASTER;
+ adc_reg &= ~AD1938_ADC_BCLK_MASTER;
+ dac_reg &= ~AD1938_DAC_LCR_MASTER;
+ dac_reg &= ~AD1938_DAC_BCLK_MASTER;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ codec->write(codec, AD1938_ADC_CTRL2, adc_reg);
+ codec->write(codec, AD1938_DAC_CTRL1, dac_reg);
+
+ return 0;
+}
+
+static int ad1938_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int word_len = 0, reg = 0;
+
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ word_len = 3;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ word_len = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S32_LE:
+ word_len = 0;
+ break;
+ }
+
+ reg = codec->read(codec, AD1938_DAC_CTRL2);
+ reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len;
+ codec->write(codec, AD1938_DAC_CTRL2, reg);
+
+ reg = codec->read(codec, AD1938_ADC_CTRL1);
+ reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len;
+ codec->write(codec, AD1938_ADC_CTRL1, reg);
+
+ return 0;
+}
+
+static int ad1938_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ ad1938_pll_powerctrl(codec, 1);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ case SND_SOC_BIAS_OFF:
+ ad1938_pll_powerctrl(codec, 0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+/*
+ * interface to read/write ad1938 register
+ */
+
+#define AD1938_SPI_ADDR 0x4
+#define AD1938_SPI_READ 0x1
+#define AD1938_SPI_BUFLEN 3
+
+/*
+ * write to the ad1938 register space
+ */
+
+static int ad1938_write_reg(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *reg_cache = codec->reg_cache;
+ int ret = 0;
+
+ if (value != reg_cache[reg]) {
+ uint8_t buf[AD1938_SPI_BUFLEN];
+ struct spi_transfer t = {
+ .tx_buf = buf,
+ .len = AD1938_SPI_BUFLEN,
+ };
+ struct spi_message m;
+
+ buf[0] = AD1938_SPI_ADDR << 1;
+ buf[1] = reg;
+ buf[2] = value;
+ spi_message_init(&m);
+ spi_message_add_tail(&t, &m);
+ ret = spi_sync(codec->control_data, &m);
+ if (ret == 0)
+ reg_cache[reg] = value;
+ }
+
+ return ret;
+}
+
+/*
+ * read from the ad1938 register space cache
+ */
+
+static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *reg_cache = codec->reg_cache;
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ return reg_cache[reg];
+}
+
+/*
+ * read from the ad1938 register space
+ */
+
+static unsigned int ad1938_read_reg(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ char w_buf[AD1938_SPI_BUFLEN];
+ char r_buf[AD1938_SPI_BUFLEN];
+ int ret;
+
+ struct spi_transfer t = {
+ .tx_buf = w_buf,
+ .rx_buf = r_buf,
+ .len = AD1938_SPI_BUFLEN,
+ };
+ struct spi_message m;
+
+ w_buf[0] = (AD1938_SPI_ADDR << 1) | AD1938_SPI_READ;
+ w_buf[1] = reg;
+ w_buf[2] = 0;
+
+ spi_message_init(&m);
+ spi_message_add_tail(&t, &m);
+ ret = spi_sync(codec->control_data, &m);
+ if (ret == 0)
+ return r_buf[2];
+ else
+ return -EIO;
+}
+
+static int ad1938_fill_cache(struct snd_soc_codec *codec)
+{
+ int i;
+ u8 *reg_cache = codec->reg_cache;
+ struct spi_device *spi = codec->control_data;
+
+ for (i = 0; i < codec->reg_cache_size; i++) {
+ int ret = ad1938_read_reg(codec, i);
+ if (ret == -EIO) {
+ dev_err(&spi->dev, "AD1938 SPI read failure\n");
+ return ret;
+ }
+ reg_cache[i] = ret;
+ }
+
+ return 0;
+}
+
+static int __devinit ad1938_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_codec *codec;
+ struct ad1938_priv *ad1938;
+
+ ad1938 = kzalloc(sizeof(struct ad1938_priv), GFP_KERNEL);
+ if (ad1938 == NULL)
+ return -ENOMEM;
+
+ codec = &ad1938->codec;
+ codec->control_data = spi;
+ codec->dev = &spi->dev;
+
+ dev_set_drvdata(&spi->dev, ad1938);
+
+ return ad1938_register(ad1938);
+}
+
+static int __devexit ad1938_spi_remove(struct spi_device *spi)
+{
+ struct ad1938_priv *ad1938 = dev_get_drvdata(&spi->dev);
+
+ ad1938_unregister(ad1938);
+ return 0;
+}
+
+static struct spi_driver ad1938_spi_driver = {
+ .driver = {
+ .name = "ad1938",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = ad1938_spi_probe,
+ .remove = __devexit_p(ad1938_spi_remove),
+};
+
+static struct snd_soc_dai_ops ad1938_dai_ops = {
+ .hw_params = ad1938_hw_params,
+ .digital_mute = ad1938_mute,
+ .set_tdm_slot = ad1938_set_tdm_slot,
+ .set_fmt = ad1938_set_dai_fmt,
+};
+
+/* codec DAI instance */
+struct snd_soc_dai ad1938_dai = {
+ .name = "AD1938",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 4,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+ },
+ .ops = &ad1938_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ad1938_dai);
+
+static int ad1938_register(struct ad1938_priv *ad1938)
+{
+ int ret;
+ struct snd_soc_codec *codec = &ad1938->codec;
+
+ if (ad1938_codec) {
+ dev_err(codec->dev, "Another ad1938 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ codec->private_data = ad1938;
+ codec->reg_cache = ad1938->reg_cache;
+ codec->reg_cache_size = AD1938_NUM_REGS;
+ codec->name = "AD1938";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ad1938_dai;
+ codec->num_dai = 1;
+ codec->write = ad1938_write_reg;
+ codec->read = ad1938_read_reg_cache;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ad1938_dai.dev = codec->dev;
+ ad1938_codec = codec;
+
+ /* default setting for ad1938 */
+
+ /* unmute dac channels */
+ codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0);
+ /* de-emphasis: 48kHz, powedown dac */
+ codec->write(codec, AD1938_DAC_CTRL2, 0x1A);
+ /* powerdown dac, dac in tdm mode */
+ codec->write(codec, AD1938_DAC_CTRL0, 0x41);
+ /* high-pass filter enable */
+ codec->write(codec, AD1938_ADC_CTRL0, 0x3);
+ /* sata delay=1, adc aux mode */
+ codec->write(codec, AD1938_ADC_CTRL1, 0x43);
+ /* pll input: mclki/xi */
+ codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D);
+ codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04);
+
+ ad1938_fill_cache(codec);
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ kfree(ad1938);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&ad1938_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ kfree(ad1938);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void ad1938_unregister(struct ad1938_priv *ad1938)
+{
+ ad1938_set_bias_level(&ad1938->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&ad1938_dai);
+ snd_soc_unregister_codec(&ad1938->codec);
+ kfree(ad1938);
+ ad1938_codec = NULL;
+}
+
+static int ad1938_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (ad1938_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ad1938_codec;
+ codec = ad1938_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, ad1938_snd_controls,
+ ARRAY_SIZE(ad1938_snd_controls));
+ snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets,
+ ARRAY_SIZE(ad1938_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_new_widgets(codec);
+
+ ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int ad1938_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int ad1938_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ ad1938_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ad1938_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ ad1938_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+#else
+#define ad1938_suspend NULL
+#define ad1938_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_ad1938 = {
+ .probe = ad1938_probe,
+ .remove = ad1938_remove,
+ .suspend = ad1938_suspend,
+ .resume = ad1938_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938);
+
+static int __init ad1938_init(void)
+{
+ int ret;
+
+ ret = spi_register_driver(&ad1938_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register ad1938 SPI driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(ad1938_init);
+
+static void __exit ad1938_exit(void)
+{
+ spi_unregister_driver(&ad1938_spi_driver);
+}
+module_exit(ad1938_exit);
+
+MODULE_DESCRIPTION("ASoC ad1938 driver");
+MODULE_AUTHOR("Barry Song ");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1938.h b/sound/soc/codecs/ad1938.h
new file mode 100644
index 0000000..fe3c48c
--- /dev/null
+++ b/sound/soc/codecs/ad1938.h
@@ -0,0 +1,100 @@
+/*
+ * File: sound/soc/codecs/ad1836.h
+ * Based on:
+ * Author: Barry Song <Barry.Song@analog.com>
+ *
+ * Created: May 25, 2009
+ * Description: definitions for AD1938 registers
+ *
+ * Modified:
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef __AD1938_H__
+#define __AD1938_H__
+
+#define AD1938_PLL_CLK_CTRL0 0
+#define AD1938_PLL_POWERDOWN 0x01
+#define AD1938_PLL_CLK_CTRL1 1
+#define AD1938_DAC_CTRL0 2
+#define AD1938_DAC_POWERDOWN 0x01
+#define AD1938_DAC_SERFMT_MASK 0xC0
+#define AD1938_DAC_SERFMT_STEREO (0 << 6)
+#define AD1938_DAC_SERFMT_TDM (1 << 6)
+#define AD1938_DAC_CTRL1 3
+#define AD1938_DAC_2_CHANNELS 0
+#define AD1938_DAC_4_CHANNELS 1
+#define AD1938_DAC_8_CHANNELS 2
+#define AD1938_DAC_16_CHANNELS 3
+#define AD1938_DAC_CHAN_SHFT 1
+#define AD1938_DAC_CHAN_MASK (3 << AD1938_DAC_CHAN_SHFT)
+#define AD1938_DAC_LCR_MASTER (1 << 4)
+#define AD1938_DAC_BCLK_MASTER (1 << 5)
+#define AD1938_DAC_LEFT_HIGH (1 << 3)
+#define AD1938_DAC_BCLK_INV (1 << 7)
+#define AD1938_DAC_CTRL2 4
+#define AD1938_DAC_WORD_LEN_MASK 0xC
+#define AD1938_DAC_MASTER_MUTE 1
+#define AD1938_DAC_CHNL_MUTE 5
+#define AD1938_DACL1_MUTE 0
+#define AD1938_DACR1_MUTE 1
+#define AD1938_DACL2_MUTE 2
+#define AD1938_DACR2_MUTE 3
+#define AD1938_DACL3_MUTE 4
+#define AD1938_DACR3_MUTE 5
+#define AD1938_DACL4_MUTE 6
+#define AD1938_DACR4_MUTE 7
+#define AD1938_DAC_L1_VOL 6
+#define AD1938_DAC_R1_VOL 7
+#define AD1938_DAC_L2_VOL 8
+#define AD1938_DAC_R2_VOL 9
+#define AD1938_DAC_L3_VOL 10
+#define AD1938_DAC_R3_VOL 11
+#define AD1938_DAC_L4_VOL 12
+#define AD1938_DAC_R4_VOL 13
+#define AD1938_ADC_CTRL0 14
+#define AD1938_ADC_POWERDOWN 0x01
+#define AD1938_ADC_HIGHPASS_FILTER 1
+#define AD1938_ADCL1_MUTE 2
+#define AD1938_ADCR1_MUTE 3
+#define AD1938_ADCL2_MUTE 4
+#define AD1938_ADCR2_MUTE 5
+#define AD1938_ADC_CTRL1 15
+#define AD1938_ADC_SERFMT_MASK 0x60
+#define AD1938_ADC_SERFMT_STEREO (0 << 5)
+#define AD1938_ADC_SERFMT_TDM (1 << 2)
+#define AD1938_ADC_SERFMT_AUX (2 << 5)
+#define AD1938_ADC_WORD_LEN_MASK 0x3
+#define AD1938_ADC_CTRL2 16
+#define AD1938_ADC_2_CHANNELS 0
+#define AD1938_ADC_4_CHANNELS 1
+#define AD1938_ADC_8_CHANNELS 2
+#define AD1938_ADC_16_CHANNELS 3
+#define AD1938_ADC_CHAN_SHFT 4
+#define AD1938_ADC_CHAN_MASK (3 << AD1938_ADC_CHAN_SHFT)
+#define AD1938_ADC_LCR_MASTER (1 << 3)
+#define AD1938_ADC_BCLK_MASTER (1 << 6)
+#define AD1938_ADC_LEFT_HIGH (1 << 2)
+#define AD1938_ADC_BCLK_INV (1 << 1)
+
+#define AD1938_NUM_REGS 17
+
+extern struct snd_soc_dai ad1938_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad1938;
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index dd33802..0abec0d 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -59,21 +59,6 @@ static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
return cache[reg];
}
-static inline unsigned int ak4535_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u8 data;
- data = reg;
-
- if (codec->hw_write(codec->control_data, &data, 1) != 1)
- return -EIO;
-
- if (codec->hw_read(codec->control_data, &data, 1) != 1)
- return -EIO;
-
- return data;
-};
-
/*
* write ak4535 register cache
*/
@@ -635,7 +620,6 @@ static int ak4535_probe(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
codec->hw_write = (hw_write_t)i2c_master_send;
- codec->hw_read = (hw_read_t)i2c_master_recv;
ret = ak4535_add_i2c_device(pdev, setup);
}
#endif
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
new file mode 100644
index 0000000..e057c7b
--- /dev/null
+++ b/sound/soc/codecs/ak4642.c
@@ -0,0 +1,502 @@
+/*
+ * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+/* ** CAUTION **
+ *
+ * This is very simple driver.
+ * It can use headphone output / stereo input only
+ *
+ * AK4642 is not tested.
+ * AK4643 is tested.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "ak4642.h"
+
+#define AK4642_VERSION "0.0.1"
+
+#define PW_MGMT1 0x00
+#define PW_MGMT2 0x01
+#define SG_SL1 0x02
+#define SG_SL2 0x03
+#define MD_CTL1 0x04
+#define MD_CTL2 0x05
+#define TIMER 0x06
+#define ALC_CTL1 0x07
+#define ALC_CTL2 0x08
+#define L_IVC 0x09
+#define L_DVC 0x0a
+#define ALC_CTL3 0x0b
+#define R_IVC 0x0c
+#define R_DVC 0x0d
+#define MD_CTL3 0x0e
+#define MD_CTL4 0x0f
+#define PW_MGMT3 0x10
+#define DF_S 0x11
+#define FIL3_0 0x12
+#define FIL3_1 0x13
+#define FIL3_2 0x14
+#define FIL3_3 0x15
+#define EQ_0 0x16
+#define EQ_1 0x17
+#define EQ_2 0x18
+#define EQ_3 0x19
+#define EQ_4 0x1a
+#define EQ_5 0x1b
+#define FIL1_0 0x1c
+#define FIL1_1 0x1d
+#define FIL1_2 0x1e
+#define FIL1_3 0x1f
+#define PW_MGMT4 0x20
+#define MD_CTL5 0x21
+#define LO_MS 0x22
+#define HP_MS 0x23
+#define SPK_MS 0x24
+
+#define AK4642_CACHEREGNUM 0x25
+
+struct snd_soc_codec_device soc_codec_dev_ak4642;
+
+/* codec private data */
+struct ak4642_priv {
+ struct snd_soc_codec codec;
+ unsigned int sysclk;
+};
+
+static struct snd_soc_codec *ak4642_codec;
+
+/*
+ * ak4642 register cache
+ */
+static const u16 ak4642_reg[AK4642_CACHEREGNUM] = {
+ 0x0000, 0x0000, 0x0001, 0x0000,
+ 0x0002, 0x0000, 0x0000, 0x0000,
+ 0x00e1, 0x00e1, 0x0018, 0x0000,
+ 0x00e1, 0x0018, 0x0011, 0x0008,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000,
+};
+
+/*
+ * read ak4642 register cache
+ */
+static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4642_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write ak4642 register cache
+ */
+static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4642_CACHEREGNUM)
+ return;
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the AK4642 register space
+ */
+static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D8 AK4642 register offset
+ * D7...D0 register data
+ */
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2) {
+ ak4642_write_reg_cache(codec, reg, value);
+ return 0;
+ } else
+ return -EIO;
+}
+
+static int ak4642_sync(struct snd_soc_codec *codec)
+{
+ u16 *cache = codec->reg_cache;
+ int i, r = 0;
+
+ for (i = 0; i < AK4642_CACHEREGNUM; i++)
+ r |= ak4642_write(codec, i, cache[i]);
+
+ return r;
+};
+
+static int ak4642_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (is_play) {
+ /*
+ * start headphone output
+ *
+ * PLL, Master Mode
+ * Audio I/F Format :MSB justified (ADC & DAC)
+ * Sampling Frequency: 44.1kHz
+ * Digital Volume: −8dB
+ * Bass Boost Level : Middle
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p97.
+ *
+ * Example code use 0x39, 0x79 value for 0x01 address,
+ * But we need MCKO (0x02) bit now
+ */
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x0f, 0x09);
+ ak4642_write(codec, 0x0e, 0x19);
+ ak4642_write(codec, 0x09, 0x91);
+ ak4642_write(codec, 0x0c, 0x91);
+ ak4642_write(codec, 0x0a, 0x28);
+ ak4642_write(codec, 0x0d, 0x28);
+ ak4642_write(codec, 0x00, 0x64);
+ ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */
+ ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */
+ } else {
+ /*
+ * start stereo input
+ *
+ * PLL Master Mode
+ * Audio I/F Format:MSB justified (ADC & DAC)
+ * Sampling Frequency:44.1kHz
+ * Pre MIC AMP:+20dB
+ * MIC Power On
+ * ALC setting:Refer to Table 35
+ * ALC bit=“1”
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p94.
+ */
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x02, 0x05);
+ ak4642_write(codec, 0x06, 0x3c);
+ ak4642_write(codec, 0x08, 0xe1);
+ ak4642_write(codec, 0x0b, 0x00);
+ ak4642_write(codec, 0x07, 0x21);
+ ak4642_write(codec, 0x00, 0x41);
+ ak4642_write(codec, 0x10, 0x01);
+ }
+
+ return 0;
+}
+
+static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+
+ if (is_play) {
+ /* stop headphone output */
+ ak4642_write(codec, 0x01, 0x3b);
+ ak4642_write(codec, 0x01, 0x0b);
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x0e, 0x11);
+ ak4642_write(codec, 0x0f, 0x08);
+ } else {
+ /* stop stereo input */
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x10, 0x00);
+ ak4642_write(codec, 0x07, 0x01);
+ }
+}
+
+static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ak4642_priv *ak4642 = codec->private_data;
+
+ ak4642->sysclk = freq;
+ return 0;
+}
+
+static struct snd_soc_dai_ops ak4642_dai_ops = {
+ .startup = ak4642_dai_startup,
+ .shutdown = ak4642_dai_shutdown,
+ .set_sysclk = ak4642_dai_set_sysclk,
+};
+
+struct snd_soc_dai ak4642_dai = {
+ .name = "AK4642",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE },
+ .ops = &ak4642_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ak4642_dai);
+
+static int ak4642_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ ak4642_sync(codec);
+ return 0;
+}
+
+/*
+ * initialise the AK4642 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ak4642_init(struct ak4642_priv *ak4642)
+{
+ struct snd_soc_codec *codec = &ak4642->codec;
+ int ret = 0;
+
+ if (ak4642_codec) {
+ dev_err(codec->dev, "Another ak4642 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = ak4642;
+ codec->name = "AK4642";
+ codec->owner = THIS_MODULE;
+ codec->read = ak4642_read_reg_cache;
+ codec->write = ak4642_write;
+ codec->dai = &ak4642_dai;
+ codec->num_dai = 1;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->reg_cache_size = ARRAY_SIZE(ak4642_reg);
+ codec->reg_cache = kmemdup(ak4642_reg,
+ sizeof(ak4642_reg), GFP_KERNEL);
+
+ if (!codec->reg_cache)
+ return -ENOMEM;
+
+ ak4642_dai.dev = codec->dev;
+ ak4642_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto reg_cache_err;
+ }
+
+ ret = snd_soc_register_dai(&ak4642_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ goto reg_cache_err;
+ }
+
+ /*
+ * clock setting
+ *
+ * Audio I/F Format: MSB justified (ADC & DAC)
+ * BICK frequency at Master Mode: 64fs
+ * Input Master Clock Select at PLL Mode: 11.2896MHz
+ * MCKO: Enable
+ * Sampling Frequency: 44.1kHz
+ *
+ * This operation came from example code of
+ * "ASAHI KASEI AK4642" (japanese) manual p89.
+ *
+ * please fix-me
+ */
+ ak4642_write(codec, 0x01, 0x08);
+ ak4642_write(codec, 0x04, 0x4a);
+ ak4642_write(codec, 0x05, 0x27);
+ ak4642_write(codec, 0x00, 0x40);
+ ak4642_write(codec, 0x01, 0x0b);
+
+ return ret;
+
+reg_cache_err:
+ kfree(codec->reg_cache);
+ codec->reg_cache = NULL;
+
+ return ret;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int ak4642_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ak4642_priv *ak4642;
+ struct snd_soc_codec *codec;
+ int ret;
+
+ ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL);
+ if (!ak4642)
+ return -ENOMEM;
+
+ codec = &ak4642->codec;
+ codec->dev = &i2c->dev;
+
+ i2c_set_clientdata(i2c, ak4642);
+ codec->control_data = i2c;
+
+ ret = ak4642_init(ak4642);
+ if (ret < 0)
+ printk(KERN_ERR "failed to initialise AK4642\n");
+
+ return ret;
+}
+
+static int ak4642_i2c_remove(struct i2c_client *client)
+{
+ struct ak4642_priv *ak4642 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_dai(&ak4642_dai);
+ snd_soc_unregister_codec(&ak4642->codec);
+ kfree(ak4642->codec.reg_cache);
+ kfree(ak4642);
+ ak4642_codec = NULL;
+
+ return 0;
+}
+
+static const struct i2c_device_id ak4642_i2c_id[] = {
+ { "ak4642", 0 },
+ { "ak4643", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
+
+static struct i2c_driver ak4642_i2c_driver = {
+ .driver = {
+ .name = "AK4642 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4642_i2c_probe,
+ .remove = ak4642_i2c_remove,
+ .id_table = ak4642_i2c_id,
+};
+
+#endif
+
+static int ak4642_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ int ret;
+
+ if (!ak4642_codec) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = ak4642_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4642: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4642: failed to register card\n");
+ goto card_err;
+ }
+
+ dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+
+}
+
+/* power down chip */
+static int ak4642_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
+ .remove = ak4642_remove,
+ .resume = ak4642_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642);
+
+static int __init ak4642_modinit(void)
+{
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&ak4642_i2c_driver);
+#endif
+ return ret;
+
+}
+module_init(ak4642_modinit);
+
+static void __exit ak4642_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&ak4642_i2c_driver);
+#endif
+
+}
+module_exit(ak4642_exit);
+
+MODULE_DESCRIPTION("Soc AK4642 driver");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h
new file mode 100644
index 0000000..e476833
--- /dev/null
+++ b/sound/soc/codecs/ak4642.h
@@ -0,0 +1,20 @@
+/*
+ * ak4642.h -- AK4642 Soc Audio driver
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on ak4535.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4642_H
+#define _AK4642_H
+
+extern struct snd_soc_dai ak4642_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4642;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index a32b822..ca1e24a 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
{
struct cs4270_private *cs4270 = i2c_get_clientdata(client);
struct snd_soc_codec *codec = &cs4270->codec;
- int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
- return snd_soc_write(codec, CS4270_PWRCTL, reg);
+ return snd_soc_suspend_device(codec->dev);
}
static int cs4270_i2c_resume(struct i2c_client *client)
{
struct cs4270_private *cs4270 = i2c_get_clientdata(client);
struct snd_soc_codec *codec = &cs4270->codec;
+
+ return snd_soc_resume_device(codec->dev);
+}
+
+static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg)
+{
+ struct snd_soc_codec *codec = cs4270_codec;
+ int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+ return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_codec *codec = cs4270_codec;
+ struct i2c_client *i2c_client = codec->control_data;
int reg;
/* In case the device was put to hard reset during sleep, we need to
@@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client)
for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
u8 val = snd_soc_read(codec, reg);
- if (i2c_smbus_write_byte_data(client, reg, val)) {
+ if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
dev_err(codec->dev, "i2c write failed\n");
return -EIO;
}
@@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client)
#else
#define cs4270_i2c_suspend NULL
#define cs4270_i2c_resume NULL
+#define cs4270_soc_suspend NULL
+#define cs4270_soc_resume NULL
#endif /* CONFIG_PM */
/*
@@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = {
*/
struct snd_soc_codec_device soc_codec_device_cs4270 = {
.probe = cs4270_probe,
- .remove = cs4270_remove
+ .remove = cs4270_remove,
+ .suspend = cs4270_soc_suspend,
+ .resume = cs4270_soc_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
new file mode 100644
index 0000000..38eac9c
--- /dev/null
+++ b/sound/soc/codecs/cx20442.c
@@ -0,0 +1,501 @@
+/*
+ * cx20442.c -- CX20442 ALSA Soc Audio driver
+ *
+ * Copyright 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/codecs/wm8400.c
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/tty.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+
+#include "cx20442.h"
+
+
+struct cx20442_priv {
+ struct snd_soc_codec codec;
+ u8 reg_cache[1];
+};
+
+#define CX20442_PM 0x0
+
+#define CX20442_TELIN 0
+#define CX20442_TELOUT 1
+#define CX20442_MIC 2
+#define CX20442_SPKOUT 3
+#define CX20442_AGC 4
+
+static const struct snd_soc_dapm_widget cx20442_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("TELOUT"),
+ SND_SOC_DAPM_OUTPUT("SPKOUT"),
+ SND_SOC_DAPM_OUTPUT("AGCOUT"),
+
+ SND_SOC_DAPM_MIXER("SPKOUT Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_PGA("TELOUT Amp", CX20442_PM, CX20442_TELOUT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SPKOUT Amp", CX20442_PM, CX20442_SPKOUT, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SPKOUT AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("TELIN Bias", CX20442_PM, CX20442_TELIN, 0),
+ SND_SOC_DAPM_MICBIAS("MIC Bias", CX20442_PM, CX20442_MIC, 0),
+
+ SND_SOC_DAPM_PGA("MIC AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0),
+
+ SND_SOC_DAPM_INPUT("TELIN"),
+ SND_SOC_DAPM_INPUT("MIC"),
+ SND_SOC_DAPM_INPUT("AGCIN"),
+};
+
+static const struct snd_soc_dapm_route cx20442_audio_map[] = {
+ {"TELOUT", NULL, "TELOUT Amp"},
+
+ {"SPKOUT", NULL, "SPKOUT Mixer"},
+ {"SPKOUT Mixer", NULL, "SPKOUT Amp"},
+
+ {"TELOUT Amp", NULL, "DAC"},
+ {"SPKOUT Amp", NULL, "DAC"},
+
+ {"SPKOUT Mixer", NULL, "SPKOUT AGC"},
+ {"SPKOUT AGC", NULL, "AGCIN"},
+
+ {"AGCOUT", NULL, "MIC AGC"},
+ {"MIC AGC", NULL, "MIC"},
+
+ {"MIC Bias", NULL, "MIC"},
+ {"Input Mixer", NULL, "MIC Bias"},
+
+ {"TELIN Bias", NULL, "TELIN"},
+ {"Input Mixer", NULL, "TELIN Bias"},
+
+ {"ADC", NULL, "Input Mixer"},
+};
+
+static int cx20442_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets,
+ ARRAY_SIZE(cx20442_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, cx20442_audio_map,
+ ARRAY_SIZE(cx20442_audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *reg_cache = codec->reg_cache;
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ return reg_cache[reg];
+}
+
+enum v253_vls {
+ V253_VLS_NONE = 0,
+ V253_VLS_T,
+ V253_VLS_L,
+ V253_VLS_LT,
+ V253_VLS_S,
+ V253_VLS_ST,
+ V253_VLS_M,
+ V253_VLS_MST,
+ V253_VLS_S1,
+ V253_VLS_S1T,
+ V253_VLS_MS1T,
+ V253_VLS_M1,
+ V253_VLS_M1ST,
+ V253_VLS_M1S1T,
+ V253_VLS_H,
+ V253_VLS_HT,
+ V253_VLS_MS,
+ V253_VLS_MS1,
+ V253_VLS_M1S,
+ V253_VLS_M1S1,
+ V253_VLS_TEST,
+};
+
+static int cx20442_pm_to_v253_vls(u8 value)
+{
+ switch (value & ~(1 << CX20442_AGC)) {
+ case 0:
+ return V253_VLS_T;
+ case (1 << CX20442_SPKOUT):
+ case (1 << CX20442_MIC):
+ case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC):
+ return V253_VLS_M1S1;
+ case (1 << CX20442_TELOUT):
+ case (1 << CX20442_TELIN):
+ case (1 << CX20442_TELOUT) | (1 << CX20442_TELIN):
+ return V253_VLS_L;
+ case (1 << CX20442_TELOUT) | (1 << CX20442_MIC):
+ return V253_VLS_NONE;
+ }
+ return -EINVAL;
+}
+static int cx20442_pm_to_v253_vsp(u8 value)
+{
+ switch (value & ~(1 << CX20442_AGC)) {
+ case (1 << CX20442_SPKOUT):
+ case (1 << CX20442_MIC):
+ case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC):
+ return (bool)(value & (1 << CX20442_AGC));
+ }
+ return (value & (1 << CX20442_AGC)) ? -EINVAL : 0;
+}
+
+static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *reg_cache = codec->reg_cache;
+ int vls, vsp, old, len;
+ char buf[18];
+
+ if (reg >= codec->reg_cache_size)
+ return -EINVAL;
+
+ /* hw_write and control_data pointers required for talking to the modem
+ * are expected to be set by the line discipline initialization code */
+ if (!codec->hw_write || !codec->control_data)
+ return -EIO;
+
+ old = reg_cache[reg];
+ reg_cache[reg] = value;
+
+ vls = cx20442_pm_to_v253_vls(value);
+ if (vls < 0)
+ return vls;
+
+ vsp = cx20442_pm_to_v253_vsp(value);
+ if (vsp < 0)
+ return vsp;
+
+ if ((vls == V253_VLS_T) ||
+ (vls == cx20442_pm_to_v253_vls(old))) {
+ if (vsp == cx20442_pm_to_v253_vsp(old))
+ return 0;
+ len = snprintf(buf, ARRAY_SIZE(buf), "at+vsp=%d\r", vsp);
+ } else if (vsp == cx20442_pm_to_v253_vsp(old))
+ len = snprintf(buf, ARRAY_SIZE(buf), "at+vls=%d\r", vls);
+ else
+ len = snprintf(buf, ARRAY_SIZE(buf),
+ "at+vls=%d;+vsp=%d\r", vls, vsp);
+
+ if (unlikely(len > (ARRAY_SIZE(buf) - 1)))
+ return -ENOMEM;
+
+ dev_dbg(codec->dev, "%s: %s\n", __func__, buf);
+ if (codec->hw_write(codec->control_data, buf, len) != len)
+ return -EIO;
+
+ return 0;
+}
+
+
+/* Moved up here as line discipline referres it during initialization */
+static struct snd_soc_codec *cx20442_codec;
+
+
+/*
+ * Line discpline related code
+ *
+ * Any of the callback functions below can be used in two ways:
+ * 1) registerd by a machine driver as one of line discipline operations,
+ * 2) called from a machine's provided line discipline callback function
+ * in case when extra machine specific code must be run as well.
+ */
+
+/* Modem init: echo off, digital speaker off, quiet off, voice mode */
+static const char *v253_init = "ate0m0q0+fclass=8\r";
+
+/* Line discipline .open() */
+static int v253_open(struct tty_struct *tty)
+{
+ struct snd_soc_codec *codec = cx20442_codec;
+ int ret, len = strlen(v253_init);
+
+ /* Doesn't make sense without write callback */
+ if (!tty->ops->write)
+ return -EINVAL;
+
+ /* Pass the codec structure address for use by other ldisc callbacks */
+ tty->disc_data = codec;
+
+ if (tty->ops->write(tty, v253_init, len) != len) {
+ ret = -EIO;
+ goto err;
+ }
+ /* Actual setup will be performed after the modem responds. */
+ return 0;
+err:
+ tty->disc_data = NULL;
+ return ret;
+}
+
+/* Line discipline .close() */
+static void v253_close(struct tty_struct *tty)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+
+ tty->disc_data = NULL;
+
+ if (!codec)
+ return;
+
+ /* Prevent the codec driver from further accessing the modem */
+ codec->hw_write = NULL;
+ codec->control_data = NULL;
+ codec->pop_time = 0;
+}
+
+/* Line discipline .hangup() */
+static int v253_hangup(struct tty_struct *tty)
+{
+ v253_close(tty);
+ return 0;
+}
+
+/* Line discipline .receive_buf() */
+static void v253_receive(struct tty_struct *tty,
+ const unsigned char *cp, char *fp, int count)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+
+ if (!codec)
+ return;
+
+ if (!codec->control_data) {
+ /* First modem response, complete setup procedure */
+
+ /* Set up codec driver access to modem controls */
+ codec->control_data = tty;
+ codec->hw_write = (hw_write_t)tty->ops->write;
+ codec->pop_time = 1;
+ }
+}
+
+/* Line discipline .write_wakeup() */
+static void v253_wakeup(struct tty_struct *tty)
+{
+}
+
+struct tty_ldisc_ops v253_ops = {
+ .magic = TTY_LDISC_MAGIC,
+ .name = "cx20442",
+ .owner = THIS_MODULE,
+ .open = v253_open,
+ .close = v253_close,
+ .hangup = v253_hangup,
+ .receive_buf = v253_receive,
+ .write_wakeup = v253_wakeup,
+};
+EXPORT_SYMBOL_GPL(v253_ops);
+
+
+/*
+ * Codec DAI
+ */
+
+struct snd_soc_dai cx20442_dai = {
+ .name = "CX20442",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+};
+EXPORT_SYMBOL_GPL(cx20442_dai);
+
+static int cx20442_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret;
+
+ if (!cx20442_codec) {
+ dev_err(&pdev->dev, "cx20442 not yet discovered\n");
+ return -ENODEV;
+ }
+ codec = cx20442_codec;
+
+ socdev->card->codec = codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ cx20442_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int cx20442_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device cx20442_codec_dev = {
+ .probe = cx20442_codec_probe,
+ .remove = cx20442_codec_remove,
+};
+EXPORT_SYMBOL_GPL(cx20442_codec_dev);
+
+static int cx20442_register(struct cx20442_priv *cx20442)
+{
+ struct snd_soc_codec *codec = &cx20442->codec;
+ int ret;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->name = "CX20442";
+ codec->owner = THIS_MODULE;
+ codec->private_data = cx20442;
+
+ codec->dai = &cx20442_dai;
+ codec->num_dai = 1;
+
+ codec->reg_cache = &cx20442->reg_cache;
+ codec->reg_cache_size = ARRAY_SIZE(cx20442->reg_cache);
+ codec->read = cx20442_read_reg_cache;
+ codec->write = cx20442_write;
+
+ codec->bias_level = SND_SOC_BIAS_OFF;
+
+ cx20442_dai.dev = codec->dev;
+
+ cx20442_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&cx20442_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ cx20442_codec = NULL;
+ kfree(cx20442);
+ return ret;
+}
+
+static void cx20442_unregister(struct cx20442_priv *cx20442)
+{
+ snd_soc_unregister_dai(&cx20442_dai);
+ snd_soc_unregister_codec(&cx20442->codec);
+
+ cx20442_codec = NULL;
+ kfree(cx20442);
+}
+
+static int cx20442_platform_probe(struct platform_device *pdev)
+{
+ struct cx20442_priv *cx20442;
+ struct snd_soc_codec *codec;
+
+ cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL);
+ if (cx20442 == NULL)
+ return -ENOMEM;
+
+ codec = &cx20442->codec;
+
+ codec->control_data = NULL;
+ codec->hw_write = NULL;
+ codec->pop_time = 0;
+
+ codec->dev = &pdev->dev;
+ platform_set_drvdata(pdev, cx20442);
+
+ return cx20442_register(cx20442);
+}
+
+static int __exit cx20442_platform_remove(struct platform_device *pdev)
+{
+ struct cx20442_priv *cx20442 = platform_get_drvdata(pdev);
+
+ cx20442_unregister(cx20442);
+ return 0;
+}
+
+static struct platform_driver cx20442_platform_driver = {
+ .driver = {
+ .name = "cx20442",
+ .owner = THIS_MODULE,
+ },
+ .probe = cx20442_platform_probe,
+ .remove = __exit_p(cx20442_platform_remove),
+};
+
+static int __init cx20442_init(void)
+{
+ return platform_driver_register(&cx20442_platform_driver);
+}
+module_init(cx20442_init);
+
+static void __exit cx20442_exit(void)
+{
+ platform_driver_unregister(&cx20442_platform_driver);
+}
+module_exit(cx20442_exit);
+
+MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver");
+MODULE_AUTHOR("Janusz Krzysztofik");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:cx20442");
diff --git a/sound/soc/codecs/cx20442.h b/sound/soc/codecs/cx20442.h
new file mode 100644
index 0000000..688a5eb
--- /dev/null
+++ b/sound/soc/codecs/cx20442.h
@@ -0,0 +1,20 @@
+/*
+ * cx20442.h -- audio driver for CX20442
+ *
+ * Copyright 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _CX20442_CODEC_H
+#define _CX20442_CODEC_H
+
+extern struct snd_soc_dai cx20442_dai;
+extern struct snd_soc_codec_device cx20442_codec_dev;
+extern struct tty_ldisc_ops v253_ops;
+
+#endif
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
new file mode 100644
index 0000000..9e7e964
--- /dev/null
+++ b/sound/soc/codecs/max9877.c
@@ -0,0 +1,308 @@
+/*
+ * max9877.c -- amp driver for max9877
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "max9877.h"
+
+static struct i2c_client *i2c;
+
+static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 };
+
+static void max9877_write_regs(void)
+{
+ unsigned int i;
+ u8 data[6];
+
+ data[0] = MAX9877_INPUT_MODE;
+ for (i = 0; i < ARRAY_SIZE(max9877_regs); i++)
+ data[i + 1] = max9877_regs[i];
+
+ if (i2c_master_send(i2c, data, 6) != 6)
+ dev_err(&i2c->dev, "i2c write failed\n");
+}
+
+static int max9877_get_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int mask = mc->max;
+ unsigned int invert = mc->invert;
+
+ ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
+
+ if (invert)
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int max9877_set_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int mask = mc->max;
+ unsigned int invert = mc->invert;
+ unsigned int val = (ucontrol->value.integer.value[0] & mask);
+
+ if (invert)
+ val = mask - val;
+
+ if (((max9877_regs[reg] >> shift) & mask) == val)
+ return 0;
+
+ max9877_regs[reg] &= ~(mask << shift);
+ max9877_regs[reg] |= val << shift;
+ max9877_write_regs();
+
+ return 1;
+}
+
+static int max9877_get_2reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ unsigned int mask = mc->max;
+
+ ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask;
+ ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask;
+
+ return 0;
+}
+
+static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ unsigned int mask = mc->max;
+ unsigned int val = (ucontrol->value.integer.value[0] & mask);
+ unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
+ unsigned int change = 1;
+
+ if (((max9877_regs[reg] >> shift) & mask) == val)
+ change = 0;
+
+ if (((max9877_regs[reg2] >> shift) & mask) == val2)
+ change = 0;
+
+ if (change) {
+ max9877_regs[reg] &= ~(mask << shift);
+ max9877_regs[reg] |= val << shift;
+ max9877_regs[reg2] &= ~(mask << shift);
+ max9877_regs[reg2] |= val2 << shift;
+ max9877_write_regs();
+ }
+
+ return change;
+}
+
+static int max9877_get_out_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK;
+
+ if (value)
+ value -= 1;
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int max9877_set_out_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = ucontrol->value.integer.value[0];
+
+ value += 1;
+
+ if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value)
+ return 0;
+
+ max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK;
+ max9877_regs[MAX9877_OUTPUT_MODE] |= value;
+ max9877_write_regs();
+ return 1;
+}
+
+static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK);
+
+ value = value >> MAX9877_OSC_OFFSET;
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = ucontrol->value.integer.value[0];
+
+ value = value << MAX9877_OSC_OFFSET;
+ if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value)
+ return 0;
+
+ max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK;
+ max9877_regs[MAX9877_OUTPUT_MODE] |= value;
+ max9877_write_regs();
+ return 1;
+}
+
+static const unsigned int max9877_pgain_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0),
+};
+
+static const unsigned int max9877_output_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0),
+ 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0),
+ 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const char *max9877_out_mode[] = {
+ "INA -> SPK",
+ "INA -> HP",
+ "INA -> SPK and HP",
+ "INB -> SPK",
+ "INB -> HP",
+ "INB -> SPK and HP",
+ "INA + INB -> SPK",
+ "INA + INB -> HP",
+ "INA + INB -> SPK and HP",
+};
+
+static const char *max9877_osc_mode[] = {
+ "1176KHz",
+ "1100KHz",
+ "700KHz",
+};
+
+static const struct soc_enum max9877_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode),
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode),
+};
+
+static const struct snd_kcontrol_new max9877_controls[] = {
+ SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume",
+ MAX9877_INPUT_MODE, 0, 2, 0,
+ max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
+ SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume",
+ MAX9877_INPUT_MODE, 2, 2, 0,
+ max9877_get_reg, max9877_set_reg, max9877_pgain_tlv),
+ SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume",
+ MAX9877_SPK_VOLUME, 0, 31, 0,
+ max9877_get_reg, max9877_set_reg, max9877_output_tlv),
+ SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume",
+ MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0,
+ max9877_get_2reg, max9877_set_2reg, max9877_output_tlv),
+ SOC_SINGLE_EXT("MAX9877 INB Stereo Switch",
+ MAX9877_INPUT_MODE, 4, 1, 1,
+ max9877_get_reg, max9877_set_reg),
+ SOC_SINGLE_EXT("MAX9877 INA Stereo Switch",
+ MAX9877_INPUT_MODE, 5, 1, 1,
+ max9877_get_reg, max9877_set_reg),
+ SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch",
+ MAX9877_INPUT_MODE, 6, 1, 0,
+ max9877_get_reg, max9877_set_reg),
+ SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch",
+ MAX9877_OUTPUT_MODE, 6, 1, 0,
+ max9877_get_reg, max9877_set_reg),
+ SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch",
+ MAX9877_OUTPUT_MODE, 7, 1, 1,
+ max9877_get_reg, max9877_set_reg),
+ SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0],
+ max9877_get_out_mode, max9877_set_out_mode),
+ SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1],
+ max9877_get_osc_mode, max9877_set_osc_mode),
+};
+
+/* This function is called from ASoC machine driver */
+int max9877_add_controls(struct snd_soc_codec *codec)
+{
+ return snd_soc_add_controls(codec, max9877_controls,
+ ARRAY_SIZE(max9877_controls));
+}
+EXPORT_SYMBOL_GPL(max9877_add_controls);
+
+static int __devinit max9877_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ i2c = client;
+
+ max9877_write_regs();
+
+ return 0;
+}
+
+static __devexit int max9877_i2c_remove(struct i2c_client *client)
+{
+ i2c = NULL;
+
+ return 0;
+}
+
+static const struct i2c_device_id max9877_i2c_id[] = {
+ { "max9877", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, max9877_i2c_id);
+
+static struct i2c_driver max9877_i2c_driver = {
+ .driver = {
+ .name = "max9877",
+ .owner = THIS_MODULE,
+ },
+ .probe = max9877_i2c_probe,
+ .remove = __devexit_p(max9877_i2c_remove),
+ .id_table = max9877_i2c_id,
+};
+
+static int __init max9877_init(void)
+{
+ return i2c_add_driver(&max9877_i2c_driver);
+}
+module_init(max9877_init);
+
+static void __exit max9877_exit(void)
+{
+ i2c_del_driver(&max9877_i2c_driver);
+}
+module_exit(max9877_exit);
+
+MODULE_DESCRIPTION("ASoC MAX9877 amp driver");
+MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h
new file mode 100644
index 0000000..6da7229
--- /dev/null
+++ b/sound/soc/codecs/max9877.h
@@ -0,0 +1,37 @@
+/*
+ * max9877.h -- amp driver for max9877
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _MAX9877_H
+#define _MAX9877_H
+
+#define MAX9877_INPUT_MODE 0x00
+#define MAX9877_SPK_VOLUME 0x01
+#define MAX9877_HPL_VOLUME 0x02
+#define MAX9877_HPR_VOLUME 0x03
+#define MAX9877_OUTPUT_MODE 0x04
+
+/* MAX9877_INPUT_MODE */
+#define MAX9877_INB (1 << 4)
+#define MAX9877_INA (1 << 5)
+#define MAX9877_ZCD (1 << 6)
+
+/* MAX9877_OUTPUT_MODE */
+#define MAX9877_OUTMODE_MASK (15 << 0)
+#define MAX9877_OSC_MASK (3 << 4)
+#define MAX9877_OSC_OFFSET 4
+#define MAX9877_BYPASS (1 << 6)
+#define MAX9877_SHDN (1 << 7)
+
+extern int max9877_add_controls(struct snd_soc_codec *codec);
+
+#endif
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
index 218b33a..a631911 100644
--- a/sound/soc/codecs/spdif_transciever.c
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -21,6 +21,8 @@
#include "spdif_transciever.h"
+MODULE_LICENSE("GPL");
+
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
@@ -34,6 +36,7 @@ struct snd_soc_dai dit_stub_dai = {
.formats = STUB_FORMATS,
},
};
+EXPORT_SYMBOL_GPL(dit_stub_dai);
static int spdif_dit_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 8ad4b7b..befc648 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
- if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
soc_ac97_ops.write(codec->ac97, reg, val);
@@ -168,7 +168,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
- if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cb0d1bf..3395cf9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -53,6 +53,7 @@
/* codec private data */
struct aic3x_priv {
+ struct snd_soc_codec codec;
unsigned int sysclk;
int master;
};
@@ -145,8 +146,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
u8 *value)
{
*value = reg & 0xff;
- if (codec->hw_read(codec->control_data, value, 1) != 1)
- return -EIO;
+
+ value[0] = i2c_smbus_read_byte_data(codec->control_data, value[0]);
aic3x_write_reg_cache(codec, reg, *value);
return 0;
@@ -1156,11 +1157,13 @@ static int aic3x_resume(struct platform_device *pdev)
* initialise the AIC3X driver
* register the mixer and dsp interfaces with the kernel
*/
-static int aic3x_init(struct snd_soc_device *socdev)
+static int aic3x_init(struct snd_soc_codec *codec)
{
- struct snd_soc_codec *codec = socdev->card->codec;
- struct aic3x_setup_data *setup = socdev->codec_data;
- int reg, ret = 0;
+ int reg;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
codec->name = "tlv320aic3x";
codec->owner = THIS_MODULE;
@@ -1177,13 +1180,6 @@ static int aic3x_init(struct snd_soc_device *socdev)
aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
aic3x_write(codec, AIC3X_RESET, SOFT_RESET);
- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
- printk(KERN_ERR "aic3x: failed to create pcms\n");
- goto pcm_err;
- }
-
/* DAC default volume and mute */
aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON);
aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON);
@@ -1250,30 +1246,51 @@ static int aic3x_init(struct snd_soc_device *socdev)
/* off, with power on */
aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- /* setup GPIO functions */
- aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
- aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
+ return 0;
+}
- snd_soc_add_controls(codec, aic3x_snd_controls,
- ARRAY_SIZE(aic3x_snd_controls));
- aic3x_add_widgets(codec);
- ret = snd_soc_init_card(socdev);
+static struct snd_soc_codec *aic3x_codec;
+
+static int aic3x_register(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = aic3x_init(codec);
if (ret < 0) {
- printk(KERN_ERR "aic3x: failed to register card\n");
- goto card_err;
+ dev_err(codec->dev, "Failed to initialise device\n");
+ return ret;
}
- return ret;
+ aic3x_codec = codec;
-card_err:
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-pcm_err:
- kfree(codec->reg_cache);
- return ret;
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&aic3x_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register dai\n");
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
}
-static struct snd_soc_device *aic3x_socdev;
+static int aic3x_unregister(struct aic3x_priv *aic3x)
+{
+ aic3x_set_bias_level(&aic3x->codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_unregister_dai(&aic3x_dai);
+ snd_soc_unregister_codec(&aic3x->codec);
+
+ kfree(aic3x);
+ aic3x_codec = NULL;
+
+ return 0;
+}
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
@@ -1288,28 +1305,36 @@ static struct snd_soc_device *aic3x_socdev;
static int aic3x_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct snd_soc_device *socdev = aic3x_socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- int ret;
+ struct snd_soc_codec *codec;
+ struct aic3x_priv *aic3x;
- i2c_set_clientdata(i2c, codec);
+ aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
+ if (aic3x == NULL) {
+ dev_err(&i2c->dev, "failed to create private data\n");
+ return -ENOMEM;
+ }
+
+ codec = &aic3x->codec;
+ codec->dev = &i2c->dev;
+ codec->private_data = aic3x;
codec->control_data = i2c;
+ codec->hw_write = (hw_write_t) i2c_master_send;
- ret = aic3x_init(socdev);
- if (ret < 0)
- printk(KERN_ERR "aic3x: failed to initialise AIC3X\n");
- return ret;
+ i2c_set_clientdata(i2c, aic3x);
+
+ return aic3x_register(codec);
}
static int aic3x_i2c_remove(struct i2c_client *client)
{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- kfree(codec->reg_cache);
- return 0;
+ struct aic3x_priv *aic3x = i2c_get_clientdata(client);
+
+ return aic3x_unregister(aic3x);
}
static const struct i2c_device_id aic3x_i2c_id[] = {
{ "tlv320aic3x", 0 },
+ { "tlv320aic33", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
@@ -1320,56 +1345,28 @@ static struct i2c_driver aic3x_i2c_driver = {
.name = "aic3x I2C Codec",
.owner = THIS_MODULE,
},
- .probe = aic3x_i2c_probe,
+ .probe = aic3x_i2c_probe,
.remove = aic3x_i2c_remove,
.id_table = aic3x_i2c_id,
};
-static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len)
+static inline void aic3x_i2c_init(void)
{
- value[0] = i2c_smbus_read_byte_data(client, value[0]);
- return (len == 1);
-}
-
-static int aic3x_add_i2c_device(struct platform_device *pdev,
- const struct aic3x_setup_data *setup)
-{
- struct i2c_board_info info;
- struct i2c_adapter *adapter;
- struct i2c_client *client;
int ret;
ret = i2c_add_driver(&aic3x_i2c_driver);
- if (ret != 0) {
- dev_err(&pdev->dev, "can't add i2c driver\n");
- return ret;
- }
-
- memset(&info, 0, sizeof(struct i2c_board_info));
- info.addr = setup->i2c_address;
- strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE);
-
- adapter = i2c_get_adapter(setup->i2c_bus);
- if (!adapter) {
- dev_err(&pdev->dev, "can't get i2c adapter %d\n",
- setup->i2c_bus);
- goto err_driver;
- }
-
- client = i2c_new_device(adapter, &info);
- i2c_put_adapter(adapter);
- if (!client) {
- dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
- (unsigned int)info.addr);
- goto err_driver;
- }
-
- return 0;
+ if (ret)
+ printk(KERN_ERR "%s: error regsitering i2c driver, %d\n",
+ __func__, ret);
+}
-err_driver:
+static inline void aic3x_i2c_exit(void)
+{
i2c_del_driver(&aic3x_i2c_driver);
- return -ENODEV;
}
+#else
+static inline void aic3x_i2c_init(void) { }
+static inline void aic3x_i2c_exit(void) { }
#endif
static int aic3x_probe(struct platform_device *pdev)
@@ -1377,43 +1374,51 @@ static int aic3x_probe(struct platform_device *pdev)
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct aic3x_setup_data *setup;
struct snd_soc_codec *codec;
- struct aic3x_priv *aic3x;
int ret = 0;
- printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION);
+ codec = aic3x_codec;
+ if (!codec) {
+ dev_err(&pdev->dev, "Codec not registered\n");
+ return -ENODEV;
+ }
+ socdev->card->codec = codec;
setup = socdev->codec_data;
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
- return -ENOMEM;
- aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
- if (aic3x == NULL) {
- kfree(codec);
- return -ENOMEM;
+ if (setup) {
+ /* setup GPIO functions */
+ aic3x_write(codec, AIC3X_GPIO1_REG,
+ (setup->gpio_func[0] & 0xf) << 4);
+ aic3x_write(codec, AIC3X_GPIO2_REG,
+ (setup->gpio_func[1] & 0xf) << 4);
}
- codec->private_data = aic3x;
- socdev->card->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
-
- aic3x_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- if (setup->i2c_address) {
- codec->hw_write = (hw_write_t) i2c_master_send;
- codec->hw_read = (hw_read_t) aic3x_i2c_read;
- ret = aic3x_add_i2c_device(pdev, setup);
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to create pcms\n");
+ goto pcm_err;
}
-#else
- /* Add other interfaces here */
-#endif
- if (ret != 0) {
- kfree(codec->private_data);
- kfree(codec);
+ snd_soc_add_controls(codec, aic3x_snd_controls,
+ ARRAY_SIZE(aic3x_snd_controls));
+
+ aic3x_add_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to register card\n");
+ goto card_err;
}
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+pcm_err:
+ kfree(codec->reg_cache);
return ret;
}
@@ -1428,12 +1433,8 @@ static int aic3x_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_unregister_device(codec->control_data);
- i2c_del_driver(&aic3x_i2c_driver);
-#endif
- kfree(codec->private_data);
- kfree(codec);
+
+ kfree(codec->reg_cache);
return 0;
}
@@ -1448,13 +1449,15 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
static int __init aic3x_modinit(void)
{
- return snd_soc_register_dai(&aic3x_dai);
+ aic3x_i2c_init();
+
+ return 0;
}
module_init(aic3x_modinit);
static void __exit aic3x_exit(void)
{
- snd_soc_unregister_dai(&aic3x_dai);
+ aic3x_i2c_exit();
}
module_exit(aic3x_exit);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index ac827e5..9af1c88 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -282,8 +282,6 @@ int aic3x_headset_detected(struct snd_soc_codec *codec);
int aic3x_button_pressed(struct snd_soc_codec *codec);
struct aic3x_setup_data {
- int i2c_bus;
- unsigned short i2c_address;
unsigned int gpio_func[2];
};
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 4dbb853..4df7c6c 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -225,55 +225,11 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
return;
if (mute) {
- /* Bypass the reg_cache and mute the volumes
- * Headset mute is done in it's own event handler
- * Things to mute: Earpiece, PreDrivL/R, CarkitL/R
- */
- reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL);
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_EAR_GAIN),
- TWL4030_REG_EAR_CTL);
-
- reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL);
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PREDL_GAIN),
- TWL4030_REG_PREDL_CTL);
- reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL);
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PREDR_GAIN),
- TWL4030_REG_PREDL_CTL);
-
- reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL);
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PRECKL_GAIN),
- TWL4030_REG_PRECKL_CTL);
- reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PRECKR_GAIN),
- TWL4030_REG_PRECKR_CTL);
-
/* Disable PLL */
reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
reg_val &= ~TWL4030_APLL_EN;
twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
} else {
- /* Restore the volumes
- * Headset mute is done in it's own event handler
- * Things to restore: Earpiece, PreDrivL/R, CarkitL/R
- */
- twl4030_write(codec, TWL4030_REG_EAR_CTL,
- twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL));
-
- twl4030_write(codec, TWL4030_REG_PREDL_CTL,
- twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL));
- twl4030_write(codec, TWL4030_REG_PREDR_CTL,
- twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL));
-
- twl4030_write(codec, TWL4030_REG_PRECKL_CTL,
- twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL));
- twl4030_write(codec, TWL4030_REG_PRECKR_CTL,
- twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL));
-
/* Enable PLL */
reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
reg_val |= TWL4030_APLL_EN;
@@ -443,16 +399,20 @@ SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
/* Left analog microphone selection */
static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
- SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
- SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
- SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
- SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+ SOC_DAPM_SINGLE("Main Mic Capture Switch",
+ TWL4030_REG_ANAMICL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Headset Mic Capture Switch",
+ TWL4030_REG_ANAMICL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AUXL Capture Switch",
+ TWL4030_REG_ANAMICL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Carkit Mic Capture Switch",
+ TWL4030_REG_ANAMICL, 3, 1, 0),
};
/* Right analog microphone selection */
static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
- SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
- SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0),
+ SOC_DAPM_SINGLE("Sub Mic Capture Switch", TWL4030_REG_ANAMICR, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUXR Capture Switch", TWL4030_REG_ANAMICR, 2, 1, 0),
};
/* TX1 L/R Analog/Digital microphone selection */
@@ -560,6 +520,41 @@ static int micpath_event(struct snd_soc_dapm_widget *w,
return 0;
}
+/*
+ * Output PGA builder:
+ * Handle the muting and unmuting of the given output (turning off the
+ * amplifier associated with the output pin)
+ * On mute bypass the reg_cache and mute the volume
+ * On unmute: restore the register content
+ * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R
+ */
+#define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \
+static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \
+ struct snd_kcontrol *kcontrol, int event) \
+{ \
+ u8 reg_val; \
+ \
+ switch (event) { \
+ case SND_SOC_DAPM_POST_PMU: \
+ twl4030_write(w->codec, reg, \
+ twl4030_read_reg_cache(w->codec, reg)); \
+ break; \
+ case SND_SOC_DAPM_POST_PMD: \
+ reg_val = twl4030_read_reg_cache(w->codec, reg); \
+ twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
+ reg_val & (~mask), \
+ reg); \
+ break; \
+ } \
+ return 0; \
+}
+
+TWL4030_OUTPUT_PGA(earpiece, TWL4030_REG_EAR_CTL, TWL4030_EAR_GAIN);
+TWL4030_OUTPUT_PGA(predrivel, TWL4030_REG_PREDL_CTL, TWL4030_PREDL_GAIN);
+TWL4030_OUTPUT_PGA(predriver, TWL4030_REG_PREDR_CTL, TWL4030_PREDR_GAIN);
+TWL4030_OUTPUT_PGA(carkitl, TWL4030_REG_PRECKL_CTL, TWL4030_PRECKL_GAIN);
+TWL4030_OUTPUT_PGA(carkitr, TWL4030_REG_PRECKR_CTL, TWL4030_PRECKR_GAIN);
+
static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
{
unsigned char hs_ctl;
@@ -620,6 +615,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
static void headset_ramp(struct snd_soc_codec *codec, int ramp)
{
+ struct snd_soc_device *socdev = codec->socdev;
+ struct twl4030_setup_data *setup = socdev->codec_data;
+
unsigned char hs_gain, hs_pop;
struct twl4030_priv *twl4030 = codec->private_data;
/* Base values for ramp delay calculation: 2^19 - 2^26 */
@@ -629,6 +627,17 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ /* Enable external mute control, this dramatically reduces
+ * the pop-noise */
+ if (setup && setup->hs_extmute) {
+ if (setup->set_hs_extmute) {
+ setup->set_hs_extmute(1);
+ } else {
+ hs_pop |= TWL4030_EXTMUTE;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ }
+ }
+
if (ramp) {
/* Headset ramp-up according to the TRM */
hs_pop |= TWL4030_VMID_EN;
@@ -636,6 +645,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
hs_pop |= TWL4030_RAMP_EN;
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ /* Wait ramp delay time + 1, so the VMID can settle */
+ mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
+ twl4030->sysclk) + 1);
} else {
/* Headset ramp-down _not_ according to
* the TRM, but in a way that it is working */
@@ -652,6 +664,16 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
hs_pop &= ~TWL4030_VMID_EN;
twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
}
+
+ /* Disable external mute */
+ if (setup && setup->hs_extmute) {
+ if (setup->set_hs_extmute) {
+ setup->set_hs_extmute(0);
+ } else {
+ hs_pop &= ~TWL4030_EXTMUTE;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ }
+ }
}
static int headsetlpga_event(struct snd_soc_dapm_widget *w,
@@ -712,7 +734,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
reg = twl4030_read_reg_cache(w->codec, m->reg);
- if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
+ /*
+ * bypass_state[0:3] - analog HiFi bypass
+ * bypass_state[4] - analog voice bypass
+ * bypass_state[5] - digital voice bypass
+ * bypass_state[6:7] - digital HiFi bypass
+ */
+ if (m->reg == TWL4030_REG_VSTPGA) {
+ /* Voice digital bypass */
+ if (reg)
+ twl4030->bypass_state |= (1 << 5);
+ else
+ twl4030->bypass_state &= ~(1 << 5);
+ } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
/* Analog bypass */
if (reg & (1 << m->shift))
twl4030->bypass_state |=
@@ -726,12 +760,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
twl4030->bypass_state |= (1 << 4);
else
twl4030->bypass_state &= ~(1 << 4);
- } else if (m->reg == TWL4030_REG_VSTPGA) {
- /* Voice digital bypass */
- if (reg)
- twl4030->bypass_state |= (1 << 5);
- else
- twl4030->bypass_state &= ~(1 << 5);
} else {
/* Digital bypass */
if (reg & (0x7 << m->shift))
@@ -924,7 +952,7 @@ static const struct soc_enum twl4030_op_modes_enum =
ARRAY_SIZE(twl4030_op_modes_texts),
twl4030_op_modes_texts);
-int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -1005,6 +1033,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
*/
static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+/* AVADC clock priority */
+static const char *twl4030_avadc_clk_priority_texts[] = {
+ "Voice high priority", "HiFi high priority"
+};
+
+static const struct soc_enum twl4030_avadc_clk_priority_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2,
+ ARRAY_SIZE(twl4030_avadc_clk_priority_texts),
+ twl4030_avadc_clk_priority_texts);
+
static const char *twl4030_rampdelay_texts[] = {
"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
@@ -1106,6 +1144,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
0, 3, 5, 0, input_gain_tlv),
+ SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum),
+
SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
@@ -1208,13 +1248,22 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_earpiece_controls[0],
ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
+ SND_SOC_DAPM_PGA_E("Earpiece PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, earpiecepga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
/* PreDrivL/R */
SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_predrivel_controls[0],
ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
+ SND_SOC_DAPM_PGA_E("PredriveL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, predrivelpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_predriver_controls[0],
ARRAY_SIZE(twl4030_dapm_predriver_controls)),
+ SND_SOC_DAPM_PGA_E("PredriveR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, predriverpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
/* HeadsetL/R */
SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_hsol_controls[0],
@@ -1232,22 +1281,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_carkitl_controls[0],
ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
+ SND_SOC_DAPM_PGA_E("CarkitL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, carkitlpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_carkitr_controls[0],
ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
+ SND_SOC_DAPM_PGA_E("CarkitR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, carkitrpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
/* Output MUX controls */
/* HandsfreeL/R */
SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_handsfreel_control),
- SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0,
+ SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_handsfreelmute_control),
SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
0, 0, NULL, 0, handsfreelpga_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
&twl4030_dapm_handsfreer_control),
- SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0,
+ SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_handsfreermute_control),
SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
0, 0, NULL, 0, handsfreerpga_event,
@@ -1282,11 +1337,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
SND_SOC_DAPM_POST_REG),
/* Analog input mixers for the capture amplifiers */
- SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+ SND_SOC_DAPM_MIXER("Analog Left",
TWL4030_REG_ANAMICL, 4, 0,
&twl4030_dapm_analoglmic_controls[0],
ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
- SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+ SND_SOC_DAPM_MIXER("Analog Right",
TWL4030_REG_ANAMICR, 4, 0,
&twl4030_dapm_analogrmic_controls[0],
ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
@@ -1326,16 +1381,19 @@ static const struct snd_soc_dapm_route intercon[] = {
{"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
{"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"Earpiece PGA", NULL, "Earpiece Mixer"},
/* PreDrivL */
{"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
{"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"PredriveL PGA", NULL, "PredriveL Mixer"},
/* PreDrivR */
{"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
{"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
{"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
{"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"PredriveR PGA", NULL, "PredriveR Mixer"},
/* HeadsetL */
{"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
@@ -1350,24 +1408,26 @@ static const struct snd_soc_dapm_route intercon[] = {
{"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
{"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
{"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"CarkitL PGA", NULL, "CarkitL Mixer"},
/* CarkitR */
{"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
{"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
{"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"CarkitR PGA", NULL, "CarkitR Mixer"},
/* HandsfreeL */
{"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
{"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
{"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
{"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
- {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"},
- {"HandsfreeL PGA", NULL, "HandsfreeL Switch"},
+ {"HandsfreeL", "Switch", "HandsfreeL Mux"},
+ {"HandsfreeL PGA", NULL, "HandsfreeL"},
/* HandsfreeR */
{"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
{"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
{"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
{"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
- {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"},
- {"HandsfreeR PGA", NULL, "HandsfreeR Switch"},
+ {"HandsfreeR", "Switch", "HandsfreeR Mux"},
+ {"HandsfreeR PGA", NULL, "HandsfreeR"},
/* Vibra */
{"Vibra Mux", "AudioL1", "DAC Left1"},
{"Vibra Mux", "AudioR1", "DAC Right1"},
@@ -1377,29 +1437,29 @@ static const struct snd_soc_dapm_route intercon[] = {
/* outputs */
{"OUTL", NULL, "Analog L2 Playback Mixer"},
{"OUTR", NULL, "Analog R2 Playback Mixer"},
- {"EARPIECE", NULL, "Earpiece Mixer"},
- {"PREDRIVEL", NULL, "PredriveL Mixer"},
- {"PREDRIVER", NULL, "PredriveR Mixer"},
+ {"EARPIECE", NULL, "Earpiece PGA"},
+ {"PREDRIVEL", NULL, "PredriveL PGA"},
+ {"PREDRIVER", NULL, "PredriveR PGA"},
{"HSOL", NULL, "HeadsetL PGA"},
{"HSOR", NULL, "HeadsetR PGA"},
- {"CARKITL", NULL, "CarkitL Mixer"},
- {"CARKITR", NULL, "CarkitR Mixer"},
+ {"CARKITL", NULL, "CarkitL PGA"},
+ {"CARKITR", NULL, "CarkitR PGA"},
{"HFL", NULL, "HandsfreeL PGA"},
{"HFR", NULL, "HandsfreeR PGA"},
{"Vibra Route", "Audio", "Vibra Mux"},
{"VIBRA", NULL, "Vibra Route"},
/* Capture path */
- {"Analog Left Capture Route", "Main mic", "MAINMIC"},
- {"Analog Left Capture Route", "Headset mic", "HSMIC"},
- {"Analog Left Capture Route", "AUXL", "AUXL"},
- {"Analog Left Capture Route", "Carkit mic", "CARKITMIC"},
+ {"Analog Left", "Main Mic Capture Switch", "MAINMIC"},
+ {"Analog Left", "Headset Mic Capture Switch", "HSMIC"},
+ {"Analog Left", "AUXL Capture Switch", "AUXL"},
+ {"Analog Left", "Carkit Mic Capture Switch", "CARKITMIC"},
- {"Analog Right Capture Route", "Sub mic", "SUBMIC"},
- {"Analog Right Capture Route", "AUXR", "AUXR"},
+ {"Analog Right", "Sub Mic Capture Switch", "SUBMIC"},
+ {"Analog Right", "AUXR Capture Switch", "AUXR"},
- {"ADC Physical Left", NULL, "Analog Left Capture Route"},
- {"ADC Physical Right", NULL, "Analog Right Capture Route"},
+ {"ADC Physical Left", NULL, "Analog Left"},
+ {"ADC Physical Right", NULL, "Analog Right"},
{"Digimic0 Enable", NULL, "DIGIMIC0"},
{"Digimic1 Enable", NULL, "DIGIMIC1"},
@@ -1423,11 +1483,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
/* Analog bypass routes */
- {"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"},
- {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
- {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
- {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
- {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
+ {"Right1 Analog Loopback", "Switch", "Analog Right"},
+ {"Left1 Analog Loopback", "Switch", "Analog Left"},
+ {"Right2 Analog Loopback", "Switch", "Analog Right"},
+ {"Left2 Analog Loopback", "Switch", "Analog Left"},
+ {"Voice Analog Loopback", "Switch", "Analog Left"},
{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
@@ -1609,8 +1669,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
/* If the substream has 4 channel, do the necessary setup */
if (params_channels(params) == 4) {
- u8 format, mode;
-
format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
@@ -1806,6 +1864,19 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
+static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+
+ if (tristate)
+ reg |= TWL4030_AIF_TRI_EN;
+ else
+ reg &= ~TWL4030_AIF_TRI_EN;
+
+ return twl4030_write(codec, TWL4030_REG_AUDIO_IF, reg);
+}
+
/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
* (VTXL, VTXR) for uplink has to be enabled/disabled. */
static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
@@ -1948,7 +2019,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFM:
+ case SND_SOC_DAIFMT_CBM_CFM:
format &= ~(TWL4030_VIF_SLAVE_EN);
break;
case SND_SOC_DAIFMT_CBS_CFS:
@@ -1980,6 +2051,19 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
+static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+
+ if (tristate)
+ reg |= TWL4030_VIF_TRI_EN;
+ else
+ reg &= ~TWL4030_VIF_TRI_EN;
+
+ return twl4030_write(codec, TWL4030_REG_VOICE_IF, reg);
+}
+
#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
@@ -1989,6 +2073,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = {
.hw_params = twl4030_hw_params,
.set_sysclk = twl4030_set_dai_sysclk,
.set_fmt = twl4030_set_dai_fmt,
+ .set_tristate = twl4030_set_tristate,
};
static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
@@ -1997,6 +2082,7 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
.hw_params = twl4030_voice_hw_params,
.set_sysclk = twl4030_voice_set_dai_sysclk,
.set_fmt = twl4030_voice_set_dai_fmt,
+ .set_tristate = twl4030_voice_set_tristate,
};
struct snd_soc_dai twl4030_dai[] = {
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index fe5f395..2b4bfa2 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -274,6 +274,8 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030;
struct twl4030_setup_data {
unsigned int ramp_delay_value;
unsigned int sysclk;
+ unsigned int hs_extmute:1;
+ void (*set_hs_extmute)(int mute);
};
#endif /* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 269b108..c33b92e 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -163,7 +163,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute)
else
mute_reg &= ~(1<<2);
- uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+ uda134x_write(codec, UDA134X_DATA010, mute_reg);
return 0;
}
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 5b21594..92ec034 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -5,9 +5,7 @@
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
- * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
- * Improved support for DAPM and audio routing/mixing capabilities,
- * added TLV support.
+ * Copyright (c) 2007-2009 Philipp Zabel <philipp.zabel@gmail.com>
*
* Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
* codec model.
@@ -19,26 +17,32 @@
#include <linux/module.h>
#include <linux/init.h>
#include <linux/types.h>
-#include <linux/string.h>
#include <linux/slab.h>
#include <linux/errno.h>
-#include <linux/ioctl.h>
+#include <linux/gpio.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/workqueue.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
-#include <sound/info.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
+#include <sound/uda1380.h>
#include "uda1380.h"
-static struct work_struct uda1380_work;
static struct snd_soc_codec *uda1380_codec;
+/* codec private data */
+struct uda1380_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[UDA1380_CACHEREGNUM];
+ unsigned int dac_clk;
+ struct work_struct work;
+};
+
/*
* uda1380 register cache
*/
@@ -473,6 +477,7 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
+ struct uda1380_priv *uda1380 = codec->private_data;
int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
switch (cmd) {
@@ -480,13 +485,13 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
uda1380_write_reg_cache(codec, UDA1380_MIXER,
mixer & ~R14_SILENCE);
- schedule_work(&uda1380_work);
+ schedule_work(&uda1380->work);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
uda1380_write_reg_cache(codec, UDA1380_MIXER,
mixer | R14_SILENCE);
- schedule_work(&uda1380_work);
+ schedule_work(&uda1380->work);
break;
}
return 0;
@@ -670,44 +675,33 @@ static int uda1380_resume(struct platform_device *pdev)
return 0;
}
-/*
- * initialise the UDA1380 driver
- * register mixer and dsp interfaces with the kernel
- */
-static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
+static int uda1380_probe(struct platform_device *pdev)
{
- struct snd_soc_codec *codec = socdev->card->codec;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct uda1380_platform_data *pdata;
int ret = 0;
- codec->name = "UDA1380";
- codec->owner = THIS_MODULE;
- codec->read = uda1380_read_reg_cache;
- codec->write = uda1380_write;
- codec->set_bias_level = uda1380_set_bias_level;
- codec->dai = uda1380_dai;
- codec->num_dai = ARRAY_SIZE(uda1380_dai);
- codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
- GFP_KERNEL);
- if (codec->reg_cache == NULL)
- return -ENOMEM;
- codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
- codec->reg_cache_step = 1;
- uda1380_reset(codec);
+ if (uda1380_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
- uda1380_codec = codec;
- INIT_WORK(&uda1380_work, uda1380_flush_work);
+ socdev->card->codec = uda1380_codec;
+ codec = uda1380_codec;
+ pdata = codec->dev->platform_data;
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
- pr_err("uda1380: failed to create pcms\n");
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
goto pcm_err;
}
/* power on device */
uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* set clock input */
- switch (dac_clk) {
+ switch (pdata->dac_clk) {
case UDA1380_DAC_CLK_SYSCLK:
uda1380_write(codec, UDA1380_CLK, 0);
break;
@@ -716,13 +710,12 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
break;
}
- /* uda1380 init */
snd_soc_add_controls(codec, uda1380_snd_controls,
ARRAY_SIZE(uda1380_snd_controls));
uda1380_add_widgets(codec);
ret = snd_soc_init_card(socdev);
if (ret < 0) {
- pr_err("uda1380: failed to register card\n");
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
goto card_err;
}
@@ -732,165 +725,201 @@ card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
- kfree(codec->reg_cache);
return ret;
}
-static struct snd_soc_device *uda1380_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-static int uda1380_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
+/* power down chip */
+static int uda1380_remove(struct platform_device *pdev)
{
- struct snd_soc_device *socdev = uda1380_socdev;
- struct uda1380_setup_data *setup = socdev->codec_data;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- int ret;
-
- i2c_set_clientdata(i2c, codec);
- codec->control_data = i2c;
- ret = uda1380_init(socdev, setup->dac_clk);
- if (ret < 0)
- pr_err("uda1380: failed to initialise UDA1380\n");
+ if (codec->control_data)
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return ret;
-}
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
-static int uda1380_i2c_remove(struct i2c_client *client)
-{
- struct snd_soc_codec *codec = i2c_get_clientdata(client);
- kfree(codec->reg_cache);
return 0;
}
-static const struct i2c_device_id uda1380_i2c_id[] = {
- { "uda1380", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
-
-static struct i2c_driver uda1380_i2c_driver = {
- .driver = {
- .name = "UDA1380 I2C Codec",
- .owner = THIS_MODULE,
- },
- .probe = uda1380_i2c_probe,
- .remove = uda1380_i2c_remove,
- .id_table = uda1380_i2c_id,
+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
+ .probe = uda1380_probe,
+ .remove = uda1380_remove,
+ .suspend = uda1380_suspend,
+ .resume = uda1380_resume,
};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
-static int uda1380_add_i2c_device(struct platform_device *pdev,
- const struct uda1380_setup_data *setup)
+static int uda1380_register(struct uda1380_priv *uda1380)
{
- struct i2c_board_info info;
- struct i2c_adapter *adapter;
- struct i2c_client *client;
- int ret;
+ int ret, i;
+ struct snd_soc_codec *codec = &uda1380->codec;
+ struct uda1380_platform_data *pdata = codec->dev->platform_data;
- ret = i2c_add_driver(&uda1380_i2c_driver);
- if (ret != 0) {
- dev_err(&pdev->dev, "can't add i2c driver\n");
- return ret;
+ if (uda1380_codec) {
+ dev_err(codec->dev, "Another UDA1380 is registered\n");
+ return -EINVAL;
+ }
+
+ if (!pdata || !pdata->gpio_power || !pdata->gpio_reset)
+ return -EINVAL;
+
+ ret = gpio_request(pdata->gpio_power, "uda1380 power");
+ if (ret)
+ goto err_out;
+ ret = gpio_request(pdata->gpio_reset, "uda1380 reset");
+ if (ret)
+ goto err_gpio;
+
+ gpio_direction_output(pdata->gpio_power, 1);
+
+ /* we may need to have the clock running here - pH5 */
+ gpio_direction_output(pdata->gpio_reset, 1);
+ udelay(5);
+ gpio_set_value(pdata->gpio_reset, 0);
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = uda1380;
+ codec->name = "UDA1380";
+ codec->owner = THIS_MODULE;
+ codec->read = uda1380_read_reg_cache;
+ codec->write = uda1380_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = uda1380_set_bias_level;
+ codec->dai = uda1380_dai;
+ codec->num_dai = ARRAY_SIZE(uda1380_dai);
+ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
+ codec->reg_cache = &uda1380->reg_cache;
+ codec->reg_cache_step = 1;
+
+ memcpy(codec->reg_cache, uda1380_reg, sizeof(uda1380_reg));
+
+ ret = uda1380_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ goto err_reset;
}
- memset(&info, 0, sizeof(struct i2c_board_info));
- info.addr = setup->i2c_address;
- strlcpy(info.type, "uda1380", I2C_NAME_SIZE);
+ INIT_WORK(&uda1380->work, uda1380_flush_work);
+
+ for (i = 0; i < ARRAY_SIZE(uda1380_dai); i++)
+ uda1380_dai[i].dev = codec->dev;
- adapter = i2c_get_adapter(setup->i2c_bus);
- if (!adapter) {
- dev_err(&pdev->dev, "can't get i2c adapter %d\n",
- setup->i2c_bus);
- goto err_driver;
+ uda1380_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err_reset;
}
- client = i2c_new_device(adapter, &info);
- i2c_put_adapter(adapter);
- if (!client) {
- dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
- (unsigned int)info.addr);
- goto err_driver;
+ ret = snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+ goto err_dai;
}
return 0;
-err_driver:
- i2c_del_driver(&uda1380_i2c_driver);
- return -ENODEV;
+err_dai:
+ snd_soc_unregister_codec(codec);
+err_reset:
+ gpio_set_value(pdata->gpio_power, 0);
+ gpio_free(pdata->gpio_reset);
+err_gpio:
+ gpio_free(pdata->gpio_power);
+err_out:
+ return ret;
}
-#endif
-static int uda1380_probe(struct platform_device *pdev)
+static void uda1380_unregister(struct uda1380_priv *uda1380)
{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct uda1380_setup_data *setup;
+ struct snd_soc_codec *codec = &uda1380->codec;
+ struct uda1380_platform_data *pdata = codec->dev->platform_data;
+
+ snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+ snd_soc_unregister_codec(&uda1380->codec);
+
+ gpio_set_value(pdata->gpio_power, 0);
+ gpio_free(pdata->gpio_reset);
+ gpio_free(pdata->gpio_power);
+
+ kfree(uda1380);
+ uda1380_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int uda1380_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct uda1380_priv *uda1380;
struct snd_soc_codec *codec;
int ret;
- setup = socdev->codec_data;
- codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (codec == NULL)
+ uda1380 = kzalloc(sizeof(struct uda1380_priv), GFP_KERNEL);
+ if (uda1380 == NULL)
return -ENOMEM;
- socdev->card->codec = codec;
- mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
+ codec = &uda1380->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
- uda1380_socdev = socdev;
- ret = -ENODEV;
+ i2c_set_clientdata(i2c, uda1380);
+ codec->control_data = i2c;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- if (setup->i2c_address) {
- codec->hw_write = (hw_write_t)i2c_master_send;
- ret = uda1380_add_i2c_device(pdev, setup);
- }
-#endif
+ codec->dev = &i2c->dev;
+ ret = uda1380_register(uda1380);
if (ret != 0)
- kfree(codec);
+ kfree(uda1380);
+
return ret;
}
-/* power down chip */
-static int uda1380_remove(struct platform_device *pdev)
+static int __devexit uda1380_i2c_remove(struct i2c_client *i2c)
{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec = socdev->card->codec;
-
- if (codec->control_data)
- uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
- snd_soc_free_pcms(socdev);
- snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_unregister_device(codec->control_data);
- i2c_del_driver(&uda1380_i2c_driver);
-#endif
- kfree(codec);
-
+ struct uda1380_priv *uda1380 = i2c_get_clientdata(i2c);
+ uda1380_unregister(uda1380);
return 0;
}
-struct snd_soc_codec_device soc_codec_dev_uda1380 = {
- .probe = uda1380_probe,
- .remove = uda1380_remove,
- .suspend = uda1380_suspend,
- .resume = uda1380_resume,
+static const struct i2c_device_id uda1380_i2c_id[] = {
+ { "uda1380", 0 },
+ { }
};
-EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
+
+static struct i2c_driver uda1380_i2c_driver = {
+ .driver = {
+ .name = "UDA1380 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = uda1380_i2c_probe,
+ .remove = __devexit_p(uda1380_i2c_remove),
+ .id_table = uda1380_i2c_id,
+};
+#endif
static int __init uda1380_modinit(void)
{
- return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&uda1380_i2c_driver);
+ if (ret != 0)
+ pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
+#endif
+ return 0;
}
module_init(uda1380_modinit);
static void __exit uda1380_exit(void)
{
- snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&uda1380_i2c_driver);
+#endif
}
module_exit(uda1380_exit);
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
index c55c17a..9cefa8a 100644
--- a/sound/soc/codecs/uda1380.h
+++ b/sound/soc/codecs/uda1380.h
@@ -72,14 +72,6 @@
#define R22_SKIP_DCFIL 0x0002
#define R23_AGC_EN 0x0001
-struct uda1380_setup_data {
- int i2c_bus;
- unsigned short i2c_address;
- int dac_clk;
-#define UDA1380_DAC_CLK_SYSCLK 0
-#define UDA1380_DAC_CLK_WSPLL 1
-};
-
#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e7348d3..3ff0373 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -63,6 +63,8 @@ struct wm8350_data {
struct wm8350_jack_data hpl;
struct wm8350_jack_data hpr;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+ int fll_freq_out;
+ int fll_freq_in;
};
static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
@@ -406,7 +408,6 @@ static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
-static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
static const char *wm8350_adcfilter[] = { "None", "High Pass" };
static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
static const char *wm8350_lr[] = { "Left", "Right" };
@@ -416,7 +417,6 @@ static const struct soc_enum wm8350_enum[] = {
SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
- SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
@@ -444,10 +444,9 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = {
0, 255, 0, dac_pcm_tlv),
SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
- SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
- SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
- SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
- SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+ SOC_ENUM("Capture PCM Filter", wm8350_enum[4]),
+ SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]),
+ SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]),
SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
WM8350_ADC_DIGITAL_VOLUME_L,
WM8350_ADC_DIGITAL_VOLUME_R,
@@ -613,7 +612,7 @@ SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
/* Out4 Capture Mux */
static const struct snd_kcontrol_new wm8350_out4_capture_controls =
-SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+SOC_DAPM_ENUM("Route", wm8350_enum[7]);
static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
@@ -993,6 +992,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8350 *wm8350 = codec->control_data;
u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
~WM8350_AIF_WL_MASK;
@@ -1012,6 +1012,19 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
}
wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+
+ /* The sloping stopband filter is recommended for use with
+ * lower sample rates to improve performance.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (params_rate(params) < 24000)
+ wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
+ WM8350_DAC_SB_FILT);
+ else
+ wm8350_clear_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
+ WM8350_DAC_SB_FILT);
+ }
+
return 0;
}
@@ -1093,10 +1106,14 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
{
struct snd_soc_codec *codec = codec_dai->codec;
struct wm8350 *wm8350 = codec->control_data;
+ struct wm8350_data *priv = codec->private_data;
struct _fll_div fll_div;
int ret = 0;
u16 fll_1, fll_4;
+ if (freq_in == priv->fll_freq_in && freq_out == priv->fll_freq_out)
+ return 0;
+
/* power down FLL - we need to do this for reconfiguration */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
@@ -1131,6 +1148,9 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+ priv->fll_freq_out = freq_out;
+ priv->fll_freq_in = freq_in;
+
return 0;
}
@@ -1660,6 +1680,21 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m)
+{
+ return snd_soc_suspend_device(&pdev->dev);
+}
+
+static int wm8350_codec_resume(struct platform_device *pdev)
+{
+ return snd_soc_resume_device(&pdev->dev);
+}
+#else
+#define wm8350_codec_suspend NULL
+#define wm8350_codec_resume NULL
+#endif
+
static struct platform_driver wm8350_codec_driver = {
.driver = {
.name = "wm8350-codec",
@@ -1667,6 +1702,8 @@ static struct platform_driver wm8350_codec_driver = {
},
.probe = wm8350_codec_probe,
.remove = __devexit_p(wm8350_codec_remove),
+ .suspend = wm8350_codec_suspend,
+ .resume = wm8350_codec_resume,
};
static __init int wm8350_init(void)
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 502eefa..b9ef4d9 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1022,10 +1022,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out)
return 0;
- if (freq_out != 0) {
+ if (freq_out) {
ret = fll_factors(wm8400, &factors, freq_in, freq_out);
if (ret != 0)
return ret;
+ } else {
+ /* Bodge GCC 4.4.0 uninitialised variable warning - it
+ * doesn't seem capable of working out that we exit if
+ * freq_out is 0 before any of the uses. */
+ memset(&factors, 0, sizeof(factors));
}
wm8400->fll_out = freq_out;
@@ -1040,7 +1045,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
reg &= ~WM8400_FLL_OSC_ENA;
wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
- if (freq_out == 0)
+ if (!freq_out)
return 0;
reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
@@ -1553,6 +1558,21 @@ static int __exit wm8400_codec_remove(struct platform_device *dev)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&pdev->dev);
+}
+
+static int wm8400_pdev_resume(struct platform_device *pdev)
+{
+ return snd_soc_resume_device(&pdev->dev);
+}
+#else
+#define wm8400_pdev_suspend NULL
+#define wm8400_pdev_resume NULL
+#endif
+
static struct platform_driver wm8400_codec_driver = {
.driver = {
.name = "wm8400-codec",
@@ -1560,6 +1580,8 @@ static struct platform_driver wm8400_codec_driver = {
},
.probe = wm8400_codec_probe,
.remove = __exit_p(wm8400_codec_remove),
+ .suspend = wm8400_pdev_suspend,
+ .resume = wm8400_pdev_resume,
};
static int __init wm8400_codec_init(void)
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index c8b8dba..060d5d0 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -58,55 +58,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
#define WM8510_POWER1_BIASEN 0x08
#define WM8510_POWER1_BUFIOEN 0x10
-/*
- * read wm8510 register cache
- */
-static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg == WM8510_RESET)
- return 0;
- if (reg >= WM8510_CACHEREGNUM)
- return -1;
- return cache[reg];
-}
-
-/*
- * write wm8510 register cache
- */
-static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg >= WM8510_CACHEREGNUM)
- return;
- cache[reg] = value;
-}
-
-/*
- * write to the WM8510 register space
- */
-static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8510 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8510_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0)
+#define wm8510_reset(c) snd_soc_write(c, WM8510_RESET, 0)
static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" };
static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
@@ -327,27 +279,27 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
if (freq_in == 0 || freq_out == 0) {
/* Clock CODEC directly from MCLK */
- reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
- wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff);
+ reg = snd_soc_read(codec, WM8510_CLOCK);
+ snd_soc_write(codec, WM8510_CLOCK, reg & 0x0ff);
/* Turn off PLL */
- reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
- wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
+ reg = snd_soc_read(codec, WM8510_POWER1);
+ snd_soc_write(codec, WM8510_POWER1, reg & 0x1df);
return 0;
}
pll_factors(freq_out*4, freq_in);
- wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
- wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
- wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff);
- wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff);
- reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
- wm8510_write(codec, WM8510_POWER1, reg | 0x020);
+ snd_soc_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
+ snd_soc_write(codec, WM8510_PLLK1, pll_div.k >> 18);
+ snd_soc_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff);
+ reg = snd_soc_read(codec, WM8510_POWER1);
+ snd_soc_write(codec, WM8510_POWER1, reg | 0x020);
/* Run CODEC from PLL instead of MCLK */
- reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
- wm8510_write(codec, WM8510_CLOCK, reg | 0x100);
+ reg = snd_soc_read(codec, WM8510_CLOCK);
+ snd_soc_write(codec, WM8510_CLOCK, reg | 0x100);
return 0;
}
@@ -363,24 +315,24 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8510_OPCLKDIV:
- reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf;
- wm8510_write(codec, WM8510_GPIO, reg | div);
+ reg = snd_soc_read(codec, WM8510_GPIO) & 0x1cf;
+ snd_soc_write(codec, WM8510_GPIO, reg | div);
break;
case WM8510_MCLKDIV:
- reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f;
- wm8510_write(codec, WM8510_CLOCK, reg | div);
+ reg = snd_soc_read(codec, WM8510_CLOCK) & 0x11f;
+ snd_soc_write(codec, WM8510_CLOCK, reg | div);
break;
case WM8510_ADCCLK:
- reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7;
- wm8510_write(codec, WM8510_ADC, reg | div);
+ reg = snd_soc_read(codec, WM8510_ADC) & 0x1f7;
+ snd_soc_write(codec, WM8510_ADC, reg | div);
break;
case WM8510_DACCLK:
- reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7;
- wm8510_write(codec, WM8510_DAC, reg | div);
+ reg = snd_soc_read(codec, WM8510_DAC) & 0x1f7;
+ snd_soc_write(codec, WM8510_DAC, reg | div);
break;
case WM8510_BCLKDIV:
- reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3;
- wm8510_write(codec, WM8510_CLOCK, reg | div);
+ reg = snd_soc_read(codec, WM8510_CLOCK) & 0x1e3;
+ snd_soc_write(codec, WM8510_CLOCK, reg | div);
break;
default:
return -EINVAL;
@@ -394,7 +346,7 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 iface = 0;
- u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
+ u16 clk = snd_soc_read(codec, WM8510_CLOCK) & 0x1fe;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -441,8 +393,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8510_write(codec, WM8510_IFACE, iface);
- wm8510_write(codec, WM8510_CLOCK, clk);
+ snd_soc_write(codec, WM8510_IFACE, iface);
+ snd_soc_write(codec, WM8510_CLOCK, clk);
return 0;
}
@@ -453,8 +405,8 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
- u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
+ u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f;
+ u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1;
/* bit size */
switch (params_format(params)) {
@@ -493,20 +445,20 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
break;
}
- wm8510_write(codec, WM8510_IFACE, iface);
- wm8510_write(codec, WM8510_ADD, adn);
+ snd_soc_write(codec, WM8510_IFACE, iface);
+ snd_soc_write(codec, WM8510_ADD, adn);
return 0;
}
static int wm8510_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
+ u16 mute_reg = snd_soc_read(codec, WM8510_DAC) & 0xffbf;
if (mute)
- wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
+ snd_soc_write(codec, WM8510_DAC, mute_reg | 0x40);
else
- wm8510_write(codec, WM8510_DAC, mute_reg);
+ snd_soc_write(codec, WM8510_DAC, mute_reg);
return 0;
}
@@ -514,13 +466,13 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
static int wm8510_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
+ u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3;
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
power1 |= 0x1; /* VMID 50k */
- wm8510_write(codec, WM8510_POWER1, power1);
+ snd_soc_write(codec, WM8510_POWER1, power1);
break;
case SND_SOC_BIAS_STANDBY:
@@ -528,18 +480,18 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
- wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
+ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3);
mdelay(100);
}
power1 |= 0x2; /* VMID 500k */
- wm8510_write(codec, WM8510_POWER1, power1);
+ snd_soc_write(codec, WM8510_POWER1, power1);
break;
case SND_SOC_BIAS_OFF:
- wm8510_write(codec, WM8510_POWER1, 0);
- wm8510_write(codec, WM8510_POWER2, 0);
- wm8510_write(codec, WM8510_POWER3, 0);
+ snd_soc_write(codec, WM8510_POWER1, 0);
+ snd_soc_write(codec, WM8510_POWER2, 0);
+ snd_soc_write(codec, WM8510_POWER3, 0);
break;
}
@@ -577,6 +529,7 @@ struct snd_soc_dai wm8510_dai = {
.rates = WM8510_RATES,
.formats = WM8510_FORMATS,},
.ops = &wm8510_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(wm8510_dai);
@@ -612,15 +565,14 @@ static int wm8510_resume(struct platform_device *pdev)
* initialise the WM8510 driver
* register the mixer and dsp interfaces with the kernel
*/
-static int wm8510_init(struct snd_soc_device *socdev)
+static int wm8510_init(struct snd_soc_device *socdev,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = socdev->card->codec;
int ret = 0;
codec->name = "WM8510";
codec->owner = THIS_MODULE;
- codec->read = wm8510_read_reg_cache;
- codec->write = wm8510_write;
codec->set_bias_level = wm8510_set_bias_level;
codec->dai = &wm8510_dai;
codec->num_dai = 1;
@@ -630,13 +582,20 @@ static int wm8510_init(struct snd_soc_device *socdev)
if (codec->reg_cache == NULL)
return -ENOMEM;
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n",
+ ret);
+ goto err;
+ }
+
wm8510_reset(codec);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "wm8510: failed to create pcms\n");
- goto pcm_err;
+ goto err;
}
/* power on device */
@@ -655,7 +614,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-pcm_err:
+err:
kfree(codec->reg_cache);
return ret;
}
@@ -678,7 +637,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- ret = wm8510_init(socdev);
+ ret = wm8510_init(socdev, SND_SOC_I2C);
if (ret < 0)
pr_err("failed to initialise WM8510\n");
@@ -758,7 +717,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi)
codec->control_data = spi;
- ret = wm8510_init(socdev);
+ ret = wm8510_init(socdev, SND_SOC_SPI);
if (ret < 0)
dev_err(&spi->dev, "failed to initialise WM8510\n");
@@ -779,30 +738,6 @@ static struct spi_driver wm8510_spi_driver = {
.probe = wm8510_spi_probe,
.remove = __devexit_p(wm8510_spi_remove),
};
-
-static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
-{
- struct spi_transfer t;
- struct spi_message m;
- u8 msg[2];
-
- if (len <= 0)
- return 0;
-
- msg[0] = data[0];
- msg[1] = data[1];
-
- spi_message_init(&m);
- memset(&t, 0, (sizeof t));
-
- t.tx_buf = &msg[0];
- t.len = len;
-
- spi_message_add_tail(&t, &m);
- spi_sync(spi, &m);
-
- return len;
-}
#endif /* CONFIG_SPI_MASTER */
static int wm8510_probe(struct platform_device *pdev)
@@ -827,13 +762,11 @@ static int wm8510_probe(struct platform_device *pdev)
wm8510_socdev = socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8510_add_i2c_device(pdev, setup);
}
#endif
#if defined(CONFIG_SPI_MASTER)
if (setup->spi) {
- codec->hw_write = (hw_write_t)wm8510_spi_write;
ret = spi_register_driver(&wm8510_spi_driver);
if (ret != 0)
printk(KERN_ERR "can't add spi driver");
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
new file mode 100644
index 0000000..25870a4
--- /dev/null
+++ b/sound/soc/codecs/wm8523.c
@@ -0,0 +1,699 @@
+/*
+ * wm8523.c -- WM8523 ALSA SoC Audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8523.h"
+
+static struct snd_soc_codec *wm8523_codec;
+struct snd_soc_codec_device soc_codec_dev_wm8523;
+
+#define WM8523_NUM_SUPPLIES 2
+static const char *wm8523_supply_names[WM8523_NUM_SUPPLIES] = {
+ "AVDD",
+ "LINEVDD",
+};
+
+#define WM8523_NUM_RATES 7
+
+/* codec private data */
+struct wm8523_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM8523_REGISTER_COUNT];
+ struct regulator_bulk_data supplies[WM8523_NUM_SUPPLIES];
+ unsigned int sysclk;
+ unsigned int rate_constraint_list[WM8523_NUM_RATES];
+ struct snd_pcm_hw_constraint_list rate_constraint;
+};
+
+static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = {
+ 0x8523, /* R0 - DEVICE_ID */
+ 0x0001, /* R1 - REVISION */
+ 0x0000, /* R2 - PSCTRL1 */
+ 0x1812, /* R3 - AIF_CTRL1 */
+ 0x0000, /* R4 - AIF_CTRL2 */
+ 0x0001, /* R5 - DAC_CTRL3 */
+ 0x0190, /* R6 - DAC_GAINL */
+ 0x0190, /* R7 - DAC_GAINR */
+ 0x0000, /* R8 - ZERO_DETECT */
+};
+
+static int wm8523_volatile_register(unsigned int reg)
+{
+ switch (reg) {
+ case WM8523_DEVICE_ID:
+ case WM8523_REVISION:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static int wm8523_reset(struct snd_soc_codec *codec)
+{
+ return snd_soc_write(codec, WM8523_DEVICE_ID, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -10000, 25, 0);
+
+static const char *wm8523_zd_count_text[] = {
+ "1024",
+ "2048",
+};
+
+static const struct soc_enum wm8523_zc_count =
+ SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text);
+
+static const struct snd_kcontrol_new wm8523_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR,
+ 0, 448, 0, dac_tlv),
+SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0),
+SOC_SINGLE("Playback Deemphasis Switch", WM8523_AIF_CTRL1, 8, 1, 0),
+SOC_DOUBLE("Playback Switch", WM8523_DAC_CTRL3, 2, 3, 1, 1),
+SOC_SINGLE("Volume Ramp Up Switch", WM8523_DAC_CTRL3, 1, 1, 0),
+SOC_SINGLE("Volume Ramp Down Switch", WM8523_DAC_CTRL3, 0, 1, 0),
+SOC_ENUM("Zero Detect Count", wm8523_zc_count),
+};
+
+static const struct snd_soc_dapm_widget wm8523_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("LINEVOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEVOUTR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ { "LINEVOUTL", NULL, "DAC" },
+ { "LINEVOUTR", NULL, "DAC" },
+};
+
+static int wm8523_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets,
+ ARRAY_SIZE(wm8523_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static struct {
+ int value;
+ int ratio;
+} lrclk_ratios[WM8523_NUM_RATES] = {
+ { 1, 128 },
+ { 2, 192 },
+ { 3, 256 },
+ { 4, 384 },
+ { 5, 512 },
+ { 6, 768 },
+ { 7, 1152 },
+};
+
+static int wm8523_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8523_priv *wm8523 = codec->private_data;
+
+ /* The set of sample rates that can be supported depends on the
+ * MCLK supplied to the CODEC - enforce this.
+ */
+ if (!wm8523->sysclk) {
+ dev_err(codec->dev,
+ "No MCLK configured, call set_sysclk() on init\n");
+ return -EINVAL;
+ }
+
+ return 0;
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &wm8523->rate_constraint);
+
+ return 0;
+}
+
+static int wm8523_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct wm8523_priv *wm8523 = codec->private_data;
+ int i;
+ u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1);
+ u16 aifctrl2 = snd_soc_read(codec, WM8523_AIF_CTRL2);
+
+ /* Find a supported LRCLK ratio */
+ for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+ if (wm8523->sysclk / params_rate(params) ==
+ lrclk_ratios[i].ratio)
+ break;
+ }
+
+ /* Should never happen, should be handled by constraints */
+ if (i == ARRAY_SIZE(lrclk_ratios)) {
+ dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n",
+ wm8523->sysclk / params_rate(params));
+ return -EINVAL;
+ }
+
+ aifctrl2 &= ~WM8523_SR_MASK;
+ aifctrl2 |= lrclk_ratios[i].value;
+
+ aifctrl1 &= ~WM8523_WL_MASK;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ aifctrl1 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ aifctrl1 |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ aifctrl1 |= 0x18;
+ break;
+ }
+
+ snd_soc_write(codec, WM8523_AIF_CTRL1, aifctrl1);
+ snd_soc_write(codec, WM8523_AIF_CTRL2, aifctrl2);
+
+ return 0;
+}
+
+static int wm8523_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8523_priv *wm8523 = codec->private_data;
+ unsigned int val;
+ int i;
+
+ wm8523->sysclk = freq;
+
+ wm8523->rate_constraint.count = 0;
+ for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+ val = freq / lrclk_ratios[i].ratio;
+ /* Check that it's a standard rate since core can't
+ * cope with others and having the odd rates confuses
+ * constraint matching.
+ */
+ switch (val) {
+ case 8000:
+ case 11025:
+ case 16000:
+ case 22050:
+ case 32000:
+ case 44100:
+ case 48000:
+ case 64000:
+ case 88200:
+ case 96000:
+ case 176400:
+ case 192000:
+ dev_dbg(codec->dev, "Supported sample rate: %dHz\n",
+ val);
+ wm8523->rate_constraint_list[i] = val;
+ wm8523->rate_constraint.count++;
+ break;
+ default:
+ dev_dbg(codec->dev, "Skipping sample rate: %dHz\n",
+ val);
+ }
+ }
+
+ /* Need at least one supported rate... */
+ if (wm8523->rate_constraint.count == 0)
+ return -EINVAL;
+
+ return 0;
+}
+
+
+static int wm8523_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1);
+
+ aifctrl1 &= ~(WM8523_BCLK_INV_MASK | WM8523_LRCLK_INV_MASK |
+ WM8523_FMT_MASK | WM8523_AIF_MSTR_MASK);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aifctrl1 |= WM8523_AIF_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ aifctrl1 |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aifctrl1 |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ aifctrl1 |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ aifctrl1 |= 0x0023;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aifctrl1 |= WM8523_BCLK_INV | WM8523_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aifctrl1 |= WM8523_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aifctrl1 |= WM8523_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, WM8523_AIF_CTRL1, aifctrl1);
+
+ return 0;
+}
+
+static int wm8523_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8523_priv *wm8523 = codec->private_data;
+ int ret, i;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* Full power on */
+ snd_soc_update_bits(codec, WM8523_PSCTRL1,
+ WM8523_SYS_ENA_MASK, 3);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies),
+ wm8523->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to enable supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ /* Initial power up */
+ snd_soc_update_bits(codec, WM8523_PSCTRL1,
+ WM8523_SYS_ENA_MASK, 1);
+
+ /* Sync back default/cached values */
+ for (i = WM8523_AIF_CTRL1;
+ i < WM8523_MAX_REGISTER; i++)
+ snd_soc_write(codec, i, wm8523->reg_cache[i]);
+
+
+ msleep(100);
+ }
+
+ /* Power up to mute */
+ snd_soc_update_bits(codec, WM8523_PSCTRL1,
+ WM8523_SYS_ENA_MASK, 2);
+
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* The chip runs through the power down sequence for us. */
+ snd_soc_update_bits(codec, WM8523_PSCTRL1,
+ WM8523_SYS_ENA_MASK, 0);
+ msleep(100);
+
+ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies),
+ wm8523->supplies);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8523_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM8523_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8523_dai_ops = {
+ .startup = wm8523_startup,
+ .hw_params = wm8523_hw_params,
+ .set_sysclk = wm8523_set_dai_sysclk,
+ .set_fmt = wm8523_set_dai_fmt,
+};
+
+struct snd_soc_dai wm8523_dai = {
+ .name = "WM8523",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2, /* Mono modes not yet supported */
+ .channels_max = 2,
+ .rates = WM8523_RATES,
+ .formats = WM8523_FORMATS,
+ },
+ .ops = &wm8523_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8523_dai);
+
+#ifdef CONFIG_PM
+static int wm8523_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8523_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define wm8523_suspend NULL
+#define wm8523_resume NULL
+#endif
+
+static int wm8523_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8523_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8523_codec;
+ codec = wm8523_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8523_snd_controls,
+ ARRAY_SIZE(wm8523_snd_controls));
+ wm8523_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8523_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8523 = {
+ .probe = wm8523_probe,
+ .remove = wm8523_remove,
+ .suspend = wm8523_suspend,
+ .resume = wm8523_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8523);
+
+static int wm8523_register(struct wm8523_priv *wm8523,
+ enum snd_soc_control_type control)
+{
+ int ret;
+ struct snd_soc_codec *codec = &wm8523->codec;
+ int i;
+
+ if (wm8523_codec) {
+ dev_err(codec->dev, "Another WM8523 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8523;
+ codec->name = "WM8523";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8523_set_bias_level;
+ codec->dai = &wm8523_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8523_REGISTER_COUNT;
+ codec->reg_cache = &wm8523->reg_cache;
+ codec->volatile_register = wm8523_volatile_register;
+
+ wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0];
+ wm8523->rate_constraint.count =
+ ARRAY_SIZE(wm8523->rate_constraint_list);
+
+ memcpy(codec->reg_cache, wm8523_reg, sizeof(wm8523_reg));
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++)
+ wm8523->supplies[i].supply = wm8523_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8523->supplies),
+ wm8523->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ goto err;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies),
+ wm8523->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_get;
+ }
+
+ ret = snd_soc_read(codec, WM8523_DEVICE_ID);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to read ID register\n");
+ goto err_enable;
+ }
+ if (ret != wm8523_reg[WM8523_DEVICE_ID]) {
+ dev_err(codec->dev, "Device is not a WM8523, ID is %x\n", ret);
+ ret = -EINVAL;
+ goto err_enable;
+ }
+
+ ret = snd_soc_read(codec, WM8523_REVISION);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to read revision register\n");
+ goto err_enable;
+ }
+ dev_info(codec->dev, "revision %c\n",
+ (ret & WM8523_CHIP_REV_MASK) + 'A');
+
+ ret = wm8523_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ goto err_enable;
+ }
+
+ wm8523_dai.dev = codec->dev;
+
+ /* Change some default settings - latch VU and enable ZC */
+ wm8523->reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU;
+ wm8523->reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC;
+
+ wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Bias level configuration will have done an extra enable */
+ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
+
+ wm8523_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8523_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+
+err_enable:
+ regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
+err_get:
+ regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
+err:
+ kfree(wm8523);
+ return ret;
+}
+
+static void wm8523_unregister(struct wm8523_priv *wm8523)
+{
+ wm8523_set_bias_level(&wm8523->codec, SND_SOC_BIAS_OFF);
+ regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
+ snd_soc_unregister_dai(&wm8523_dai);
+ snd_soc_unregister_codec(&wm8523->codec);
+ kfree(wm8523);
+ wm8523_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8523_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8523_priv *wm8523;
+ struct snd_soc_codec *codec;
+
+ wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL);
+ if (wm8523 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8523->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8523);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8523_register(wm8523, SND_SOC_I2C);
+}
+
+static __devexit int wm8523_i2c_remove(struct i2c_client *client)
+{
+ struct wm8523_priv *wm8523 = i2c_get_clientdata(client);
+ wm8523_unregister(wm8523);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&i2c->dev);
+}
+
+static int wm8523_i2c_resume(struct i2c_client *i2c)
+{
+ return snd_soc_resume_device(&i2c->dev);
+}
+#else
+#define wm8523_i2c_suspend NULL
+#define wm8523_i2c_resume NULL
+#endif
+
+static const struct i2c_device_id wm8523_i2c_id[] = {
+ { "wm8523", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id);
+
+static struct i2c_driver wm8523_i2c_driver = {
+ .driver = {
+ .name = "WM8523",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8523_i2c_probe,
+ .remove = __devexit_p(wm8523_i2c_remove),
+ .suspend = wm8523_i2c_suspend,
+ .resume = wm8523_i2c_resume,
+ .id_table = wm8523_i2c_id,
+};
+#endif
+
+static int __init wm8523_modinit(void)
+{
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8523_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n",
+ ret);
+ }
+#endif
+ return 0;
+}
+module_init(wm8523_modinit);
+
+static void __exit wm8523_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8523_i2c_driver);
+#endif
+}
+module_exit(wm8523_exit);
+
+MODULE_DESCRIPTION("ASoC WM8523 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8523.h b/sound/soc/codecs/wm8523.h
new file mode 100644
index 0000000..1aa9ce3
--- /dev/null
+++ b/sound/soc/codecs/wm8523.h
@@ -0,0 +1,160 @@
+/*
+ * wm8523.h -- WM8423 ASoC driver
+ *
+ * Copyright 2009 Wolfson Microelectronics, plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8523_H
+#define _WM8523_H
+
+/*
+ * Register values.
+ */
+#define WM8523_DEVICE_ID 0x00
+#define WM8523_REVISION 0x01
+#define WM8523_PSCTRL1 0x02
+#define WM8523_AIF_CTRL1 0x03
+#define WM8523_AIF_CTRL2 0x04
+#define WM8523_DAC_CTRL3 0x05
+#define WM8523_DAC_GAINL 0x06
+#define WM8523_DAC_GAINR 0x07
+#define WM8523_ZERO_DETECT 0x08
+
+#define WM8523_REGISTER_COUNT 9
+#define WM8523_MAX_REGISTER 0x08
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - DEVICE_ID
+ */
+#define WM8523_CHIP_ID_MASK 0xFFFF /* CHIP_ID - [15:0] */
+#define WM8523_CHIP_ID_SHIFT 0 /* CHIP_ID - [15:0] */
+#define WM8523_CHIP_ID_WIDTH 16 /* CHIP_ID - [15:0] */
+
+/*
+ * R1 (0x01) - REVISION
+ */
+#define WM8523_CHIP_REV_MASK 0x0007 /* CHIP_REV - [2:0] */
+#define WM8523_CHIP_REV_SHIFT 0 /* CHIP_REV - [2:0] */
+#define WM8523_CHIP_REV_WIDTH 3 /* CHIP_REV - [2:0] */
+
+/*
+ * R2 (0x02) - PSCTRL1
+ */
+#define WM8523_SYS_ENA_MASK 0x0003 /* SYS_ENA - [1:0] */
+#define WM8523_SYS_ENA_SHIFT 0 /* SYS_ENA - [1:0] */
+#define WM8523_SYS_ENA_WIDTH 2 /* SYS_ENA - [1:0] */
+
+/*
+ * R3 (0x03) - AIF_CTRL1
+ */
+#define WM8523_TDM_MODE_MASK 0x1800 /* TDM_MODE - [12:11] */
+#define WM8523_TDM_MODE_SHIFT 11 /* TDM_MODE - [12:11] */
+#define WM8523_TDM_MODE_WIDTH 2 /* TDM_MODE - [12:11] */
+#define WM8523_TDM_SLOT_MASK 0x0600 /* TDM_SLOT - [10:9] */
+#define WM8523_TDM_SLOT_SHIFT 9 /* TDM_SLOT - [10:9] */
+#define WM8523_TDM_SLOT_WIDTH 2 /* TDM_SLOT - [10:9] */
+#define WM8523_DEEMPH 0x0100 /* DEEMPH */
+#define WM8523_DEEMPH_MASK 0x0100 /* DEEMPH */
+#define WM8523_DEEMPH_SHIFT 8 /* DEEMPH */
+#define WM8523_DEEMPH_WIDTH 1 /* DEEMPH */
+#define WM8523_AIF_MSTR 0x0080 /* AIF_MSTR */
+#define WM8523_AIF_MSTR_MASK 0x0080 /* AIF_MSTR */
+#define WM8523_AIF_MSTR_SHIFT 7 /* AIF_MSTR */
+#define WM8523_AIF_MSTR_WIDTH 1 /* AIF_MSTR */
+#define WM8523_LRCLK_INV 0x0040 /* LRCLK_INV */
+#define WM8523_LRCLK_INV_MASK 0x0040 /* LRCLK_INV */
+#define WM8523_LRCLK_INV_SHIFT 6 /* LRCLK_INV */
+#define WM8523_LRCLK_INV_WIDTH 1 /* LRCLK_INV */
+#define WM8523_BCLK_INV 0x0020 /* BCLK_INV */
+#define WM8523_BCLK_INV_MASK 0x0020 /* BCLK_INV */
+#define WM8523_BCLK_INV_SHIFT 5 /* BCLK_INV */
+#define WM8523_BCLK_INV_WIDTH 1 /* BCLK_INV */
+#define WM8523_WL_MASK 0x0018 /* WL - [4:3] */
+#define WM8523_WL_SHIFT 3 /* WL - [4:3] */
+#define WM8523_WL_WIDTH 2 /* WL - [4:3] */
+#define WM8523_FMT_MASK 0x0007 /* FMT - [2:0] */
+#define WM8523_FMT_SHIFT 0 /* FMT - [2:0] */
+#define WM8523_FMT_WIDTH 3 /* FMT - [2:0] */
+
+/*
+ * R4 (0x04) - AIF_CTRL2
+ */
+#define WM8523_DAC_OP_MUX_MASK 0x00C0 /* DAC_OP_MUX - [7:6] */
+#define WM8523_DAC_OP_MUX_SHIFT 6 /* DAC_OP_MUX - [7:6] */
+#define WM8523_DAC_OP_MUX_WIDTH 2 /* DAC_OP_MUX - [7:6] */
+#define WM8523_BCLKDIV_MASK 0x0038 /* BCLKDIV - [5:3] */
+#define WM8523_BCLKDIV_SHIFT 3 /* BCLKDIV - [5:3] */
+#define WM8523_BCLKDIV_WIDTH 3 /* BCLKDIV - [5:3] */
+#define WM8523_SR_MASK 0x0007 /* SR - [2:0] */
+#define WM8523_SR_SHIFT 0 /* SR - [2:0] */
+#define WM8523_SR_WIDTH 3 /* SR - [2:0] */
+
+/*
+ * R5 (0x05) - DAC_CTRL3
+ */
+#define WM8523_ZC 0x0010 /* ZC */
+#define WM8523_ZC_MASK 0x0010 /* ZC */
+#define WM8523_ZC_SHIFT 4 /* ZC */
+#define WM8523_ZC_WIDTH 1 /* ZC */
+#define WM8523_DACR 0x0008 /* DACR */
+#define WM8523_DACR_MASK 0x0008 /* DACR */
+#define WM8523_DACR_SHIFT 3 /* DACR */
+#define WM8523_DACR_WIDTH 1 /* DACR */
+#define WM8523_DACL 0x0004 /* DACL */
+#define WM8523_DACL_MASK 0x0004 /* DACL */
+#define WM8523_DACL_SHIFT 2 /* DACL */
+#define WM8523_DACL_WIDTH 1 /* DACL */
+#define WM8523_VOL_UP_RAMP 0x0002 /* VOL_UP_RAMP */
+#define WM8523_VOL_UP_RAMP_MASK 0x0002 /* VOL_UP_RAMP */
+#define WM8523_VOL_UP_RAMP_SHIFT 1 /* VOL_UP_RAMP */
+#define WM8523_VOL_UP_RAMP_WIDTH 1 /* VOL_UP_RAMP */
+#define WM8523_VOL_DOWN_RAMP 0x0001 /* VOL_DOWN_RAMP */
+#define WM8523_VOL_DOWN_RAMP_MASK 0x0001 /* VOL_DOWN_RAMP */
+#define WM8523_VOL_DOWN_RAMP_SHIFT 0 /* VOL_DOWN_RAMP */
+#define WM8523_VOL_DOWN_RAMP_WIDTH 1 /* VOL_DOWN_RAMP */
+
+/*
+ * R6 (0x06) - DAC_GAINL
+ */
+#define WM8523_DACL_VU 0x0200 /* DACL_VU */
+#define WM8523_DACL_VU_MASK 0x0200 /* DACL_VU */
+#define WM8523_DACL_VU_SHIFT 9 /* DACL_VU */
+#define WM8523_DACL_VU_WIDTH 1 /* DACL_VU */
+#define WM8523_DACL_VOL_MASK 0x01FF /* DACL_VOL - [8:0] */
+#define WM8523_DACL_VOL_SHIFT 0 /* DACL_VOL - [8:0] */
+#define WM8523_DACL_VOL_WIDTH 9 /* DACL_VOL - [8:0] */
+
+/*
+ * R7 (0x07) - DAC_GAINR
+ */
+#define WM8523_DACR_VU 0x0200 /* DACR_VU */
+#define WM8523_DACR_VU_MASK 0x0200 /* DACR_VU */
+#define WM8523_DACR_VU_SHIFT 9 /* DACR_VU */
+#define WM8523_DACR_VU_WIDTH 1 /* DACR_VU */
+#define WM8523_DACR_VOL_MASK 0x01FF /* DACR_VOL - [8:0] */
+#define WM8523_DACR_VOL_SHIFT 0 /* DACR_VOL - [8:0] */
+#define WM8523_DACR_VOL_WIDTH 9 /* DACR_VOL - [8:0] */
+
+/*
+ * R8 (0x08) - ZERO_DETECT
+ */
+#define WM8523_ZD_COUNT_MASK 0x0003 /* ZD_COUNT - [1:0] */
+#define WM8523_ZD_COUNT_SHIFT 0 /* ZD_COUNT - [1:0] */
+#define WM8523_ZD_COUNT_WIDTH 2 /* ZD_COUNT - [1:0] */
+
+extern struct snd_soc_dai wm8523_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8523;
+
+#endif
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 86c4b24d..6bded8c 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -24,6 +24,8 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -187,82 +189,22 @@ struct pll_state {
unsigned int out;
};
+#define WM8580_NUM_SUPPLIES 3
+static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = {
+ "AVDD",
+ "DVDD",
+ "PVDD",
+};
+
/* codec private data */
struct wm8580_priv {
struct snd_soc_codec codec;
+ struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES];
u16 reg_cache[WM8580_MAX_REGISTER + 1];
struct pll_state a;
struct pll_state b;
};
-
-/*
- * read wm8580 register cache
- */
-static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
- return cache[reg];
-}
-
-/*
- * write wm8580 register cache
- */
-static inline void wm8580_write_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
-
- cache[reg] = value;
-}
-
-/*
- * write to the WM8580 register space
- */
-static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
-
- /* Registers are 9 bits wide */
- value &= 0x1ff;
-
- switch (reg) {
- case WM8580_RESET:
- /* Uncached */
- break;
- default:
- if (value == wm8580_read_reg_cache(codec, reg))
- return 0;
- }
-
- /* data is
- * D15..D9 WM8580 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8580_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-static inline unsigned int wm8580_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- switch (reg) {
- default:
- return wm8580_read_reg_cache(codec, reg);
- }
-}
-
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
@@ -271,25 +213,22 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 *reg_cache = codec->reg_cache;
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
int ret;
- u16 val;
/* Clear the register cache so we write without VU set */
- wm8580_write_reg_cache(codec, reg, 0);
- wm8580_write_reg_cache(codec, reg2, 0);
+ reg_cache[reg] = 0;
+ reg_cache[reg2] = 0;
ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
if (ret < 0)
return ret;
/* Now write again with the volume update bit set */
- val = wm8580_read_reg_cache(codec, reg);
- wm8580_write(codec, reg, val | 0x0100);
-
- val = wm8580_read_reg_cache(codec, reg2);
- wm8580_write(codec, reg2, val | 0x0100);
+ snd_soc_update_bits(codec, reg, 0x100, 0x100);
+ snd_soc_update_bits(codec, reg2, 0x100, 0x100);
return 0;
}
@@ -512,27 +451,27 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
/* Always disable the PLL - it is not safe to leave it running
* while reprogramming it.
*/
- reg = wm8580_read(codec, WM8580_PWRDN2);
- wm8580_write(codec, WM8580_PWRDN2, reg | pwr_mask);
+ reg = snd_soc_read(codec, WM8580_PWRDN2);
+ snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask);
if (!freq_in || !freq_out)
return 0;
- wm8580_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff);
- wm8580_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff);
- wm8580_write(codec, WM8580_PLLA3 + offset,
+ snd_soc_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8580_PLLA3 + offset,
(pll_div.k >> 18 & 0xf) | (pll_div.n << 4));
- reg = wm8580_read(codec, WM8580_PLLA4 + offset);
- reg &= ~0x3f;
+ reg = snd_soc_read(codec, WM8580_PLLA4 + offset);
+ reg &= ~0x1b;
reg |= pll_div.prescale | pll_div.postscale << 1 |
pll_div.freqmode << 3;
- wm8580_write(codec, WM8580_PLLA4 + offset, reg);
+ snd_soc_write(codec, WM8580_PLLA4 + offset, reg);
/* All done, turn it on */
- reg = wm8580_read(codec, WM8580_PWRDN2);
- wm8580_write(codec, WM8580_PWRDN2, reg & ~pwr_mask);
+ reg = snd_soc_read(codec, WM8580_PWRDN2);
+ snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask);
return 0;
}
@@ -547,7 +486,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
+ u16 paifb = snd_soc_read(codec, WM8580_PAIF3 + dai->id);
paifb &= ~WM8580_AIF_LENGTH_MASK;
/* bit size */
@@ -567,7 +506,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
+ snd_soc_write(codec, WM8580_PAIF3 + dai->id, paifb);
return 0;
}
@@ -579,8 +518,8 @@ static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int aifb;
int can_invert_lrclk;
- aifa = wm8580_read(codec, WM8580_PAIF1 + codec_dai->id);
- aifb = wm8580_read(codec, WM8580_PAIF3 + codec_dai->id);
+ aifa = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id);
+ aifb = snd_soc_read(codec, WM8580_PAIF3 + codec_dai->id);
aifb &= ~(WM8580_AIF_FMT_MASK | WM8580_AIF_LRP | WM8580_AIF_BCP);
@@ -646,8 +585,8 @@ static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8580_write(codec, WM8580_PAIF1 + codec_dai->id, aifa);
- wm8580_write(codec, WM8580_PAIF3 + codec_dai->id, aifb);
+ snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, aifa);
+ snd_soc_write(codec, WM8580_PAIF3 + codec_dai->id, aifb);
return 0;
}
@@ -660,7 +599,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8580_MCLK:
- reg = wm8580_read(codec, WM8580_PLLB4);
+ reg = snd_soc_read(codec, WM8580_PLLB4);
reg &= ~WM8580_PLLB4_MCLKOUTSRC_MASK;
switch (div) {
@@ -682,11 +621,11 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
default:
return -EINVAL;
}
- wm8580_write(codec, WM8580_PLLB4, reg);
+ snd_soc_write(codec, WM8580_PLLB4, reg);
break;
case WM8580_DAC_CLKSEL:
- reg = wm8580_read(codec, WM8580_CLKSEL);
+ reg = snd_soc_read(codec, WM8580_CLKSEL);
reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK;
switch (div) {
@@ -704,11 +643,11 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
default:
return -EINVAL;
}
- wm8580_write(codec, WM8580_CLKSEL, reg);
+ snd_soc_write(codec, WM8580_CLKSEL, reg);
break;
case WM8580_CLKOUTSRC:
- reg = wm8580_read(codec, WM8580_PLLB4);
+ reg = snd_soc_read(codec, WM8580_PLLB4);
reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK;
switch (div) {
@@ -730,7 +669,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
default:
return -EINVAL;
}
- wm8580_write(codec, WM8580_PLLB4, reg);
+ snd_soc_write(codec, WM8580_PLLB4, reg);
break;
default:
@@ -745,14 +684,14 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute)
struct snd_soc_codec *codec = codec_dai->codec;
unsigned int reg;
- reg = wm8580_read(codec, WM8580_DAC_CONTROL5);
+ reg = snd_soc_read(codec, WM8580_DAC_CONTROL5);
if (mute)
reg |= WM8580_DAC_CONTROL5_MUTEALL;
else
reg &= ~WM8580_DAC_CONTROL5_MUTEALL;
- wm8580_write(codec, WM8580_DAC_CONTROL5, reg);
+ snd_soc_write(codec, WM8580_DAC_CONTROL5, reg);
return 0;
}
@@ -769,20 +708,20 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Power up and get individual control of the DACs */
- reg = wm8580_read(codec, WM8580_PWRDN1);
+ reg = snd_soc_read(codec, WM8580_PWRDN1);
reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
- wm8580_write(codec, WM8580_PWRDN1, reg);
+ snd_soc_write(codec, WM8580_PWRDN1, reg);
/* Make VMID high impedence */
- reg = wm8580_read(codec, WM8580_ADC_CONTROL1);
+ reg = snd_soc_read(codec, WM8580_ADC_CONTROL1);
reg &= ~0x100;
- wm8580_write(codec, WM8580_ADC_CONTROL1, reg);
+ snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
}
break;
case SND_SOC_BIAS_OFF:
- reg = wm8580_read(codec, WM8580_PWRDN1);
- wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
+ reg = snd_soc_read(codec, WM8580_PWRDN1);
+ snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
break;
}
codec->bias_level = level;
@@ -893,7 +832,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
-static int wm8580_register(struct wm8580_priv *wm8580)
+static int wm8580_register(struct wm8580_priv *wm8580,
+ enum snd_soc_control_type control)
{
int ret, i;
struct snd_soc_codec *codec = &wm8580->codec;
@@ -911,8 +851,6 @@ static int wm8580_register(struct wm8580_priv *wm8580)
codec->private_data = wm8580;
codec->name = "WM8580";
codec->owner = THIS_MODULE;
- codec->read = wm8580_read_reg_cache;
- codec->write = wm8580_write;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8580_set_bias_level;
codec->dai = wm8580_dai;
@@ -922,11 +860,34 @@ static int wm8580_register(struct wm8580_priv *wm8580)
memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg));
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++)
+ wm8580->supplies[i].supply = wm8580_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8580->supplies),
+ wm8580->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ goto err;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies),
+ wm8580->supplies);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_regulator_get;
+ }
+
/* Get the codec into a known state */
- ret = wm8580_write(codec, WM8580_RESET, 0);
+ ret = snd_soc_write(codec, WM8580_RESET, 0);
if (ret != 0) {
dev_err(codec->dev, "Failed to reset codec: %d\n", ret);
- goto err;
+ goto err_regulator_enable;
}
for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++)
@@ -939,7 +900,7 @@ static int wm8580_register(struct wm8580_priv *wm8580)
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
- goto err;
+ goto err_regulator_enable;
}
ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
@@ -952,6 +913,10 @@ static int wm8580_register(struct wm8580_priv *wm8580)
err_codec:
snd_soc_unregister_codec(codec);
+err_regulator_enable:
+ regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
+err_regulator_get:
+ regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
err:
kfree(wm8580);
return ret;
@@ -962,6 +927,8 @@ static void wm8580_unregister(struct wm8580_priv *wm8580)
wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF);
snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
snd_soc_unregister_codec(&wm8580->codec);
+ regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
+ regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
kfree(wm8580);
wm8580_codec = NULL;
}
@@ -978,14 +945,13 @@ static int wm8580_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
codec = &wm8580->codec;
- codec->hw_write = (hw_write_t)i2c_master_send;
i2c_set_clientdata(i2c, wm8580);
codec->control_data = i2c;
codec->dev = &i2c->dev;
- return wm8580_register(wm8580);
+ return wm8580_register(wm8580, SND_SOC_I2C);
}
static int wm8580_i2c_remove(struct i2c_client *client)
@@ -995,6 +961,21 @@ static int wm8580_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8580_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8580_i2c_suspend NULL
+#define wm8580_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8580_i2c_id[] = {
{ "wm8580", 0 },
{ }
@@ -1008,6 +989,8 @@ static struct i2c_driver wm8580_i2c_driver = {
},
.probe = wm8580_i2c_probe,
.remove = wm8580_i2c_remove,
+ .suspend = wm8580_i2c_suspend,
+ .resume = wm8580_i2c_resume,
.id_table = wm8580_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index e7ff212..16e969a 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -43,45 +43,6 @@ static const u16 wm8728_reg_defaults[] = {
0x100,
};
-static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
- return cache[reg];
-}
-
-static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
- cache[reg] = value;
-}
-
-/*
- * write to the WM8728 register space
- */
-static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8728 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8728_write_reg_cache(codec, reg, value);
-
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1);
static const struct snd_kcontrol_new wm8728_snd_controls[] = {
@@ -121,12 +82,12 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec)
static int wm8728_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ u16 mute_reg = snd_soc_read(codec, WM8728_DACCTL);
if (mute)
- wm8728_write(codec, WM8728_DACCTL, mute_reg | 1);
+ snd_soc_write(codec, WM8728_DACCTL, mute_reg | 1);
else
- wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1);
+ snd_soc_write(codec, WM8728_DACCTL, mute_reg & ~1);
return 0;
}
@@ -138,7 +99,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+ u16 dac = snd_soc_read(codec, WM8728_DACCTL);
dac &= ~0x18;
@@ -155,7 +116,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- wm8728_write(codec, WM8728_DACCTL, dac);
+ snd_soc_write(codec, WM8728_DACCTL, dac);
return 0;
}
@@ -164,7 +125,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL);
+ u16 iface = snd_soc_read(codec, WM8728_IFCTL);
/* Currently only I2S is supported by the driver, though the
* hardware is more flexible.
@@ -204,7 +165,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8728_write(codec, WM8728_IFCTL, iface);
+ snd_soc_write(codec, WM8728_IFCTL, iface);
return 0;
}
@@ -220,19 +181,19 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Power everything up... */
- reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
- wm8728_write(codec, WM8728_DACCTL, reg & ~0x4);
+ reg = snd_soc_read(codec, WM8728_DACCTL);
+ snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4);
/* ..then sync in the register cache. */
for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++)
- wm8728_write(codec, i,
- wm8728_read_reg_cache(codec, i));
+ snd_soc_write(codec, i,
+ snd_soc_read(codec, i));
}
break;
case SND_SOC_BIAS_OFF:
- reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
- wm8728_write(codec, WM8728_DACCTL, reg | 0x4);
+ reg = snd_soc_read(codec, WM8728_DACCTL);
+ snd_soc_write(codec, WM8728_DACCTL, reg | 0x4);
break;
}
codec->bias_level = level;
@@ -287,15 +248,14 @@ static int wm8728_resume(struct platform_device *pdev)
* initialise the WM8728 driver
* register the mixer and dsp interfaces with the kernel
*/
-static int wm8728_init(struct snd_soc_device *socdev)
+static int wm8728_init(struct snd_soc_device *socdev,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = socdev->card->codec;
int ret = 0;
codec->name = "WM8728";
codec->owner = THIS_MODULE;
- codec->read = wm8728_read_reg_cache;
- codec->write = wm8728_write;
codec->set_bias_level = wm8728_set_bias_level;
codec->dai = &wm8728_dai;
codec->num_dai = 1;
@@ -307,11 +267,18 @@ static int wm8728_init(struct snd_soc_device *socdev)
if (codec->reg_cache == NULL)
return -ENOMEM;
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n",
+ ret);
+ goto err;
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "wm8728: failed to create pcms\n");
- goto pcm_err;
+ goto err;
}
/* power on device */
@@ -331,7 +298,7 @@ static int wm8728_init(struct snd_soc_device *socdev)
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-pcm_err:
+err:
kfree(codec->reg_cache);
return ret;
}
@@ -357,7 +324,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- ret = wm8728_init(socdev);
+ ret = wm8728_init(socdev, SND_SOC_I2C);
if (ret < 0)
pr_err("failed to initialise WM8728\n");
@@ -437,7 +404,7 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi)
codec->control_data = spi;
- ret = wm8728_init(socdev);
+ ret = wm8728_init(socdev, SND_SOC_SPI);
if (ret < 0)
dev_err(&spi->dev, "failed to initialise WM8728\n");
@@ -458,30 +425,6 @@ static struct spi_driver wm8728_spi_driver = {
.probe = wm8728_spi_probe,
.remove = __devexit_p(wm8728_spi_remove),
};
-
-static int wm8728_spi_write(struct spi_device *spi, const char *data, int len)
-{
- struct spi_transfer t;
- struct spi_message m;
- u8 msg[2];
-
- if (len <= 0)
- return 0;
-
- msg[0] = data[0];
- msg[1] = data[1];
-
- spi_message_init(&m);
- memset(&t, 0, (sizeof t));
-
- t.tx_buf = &msg[0];
- t.len = len;
-
- spi_message_add_tail(&t, &m);
- spi_sync(spi, &m);
-
- return len;
-}
#endif /* CONFIG_SPI_MASTER */
static int wm8728_probe(struct platform_device *pdev)
@@ -506,13 +449,11 @@ static int wm8728_probe(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8728_add_i2c_device(pdev, setup);
}
#endif
#if defined(CONFIG_SPI_MASTER)
if (setup->spi) {
- codec->hw_write = (hw_write_t)wm8728_spi_write;
ret = spi_register_driver(&wm8728_spi_driver);
if (ret != 0)
printk(KERN_ERR "can't add spi driver");
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7a20587..d3fd4f2 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -26,6 +26,7 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
+#include <sound/tlv.h>
#include "wm8731.h"
@@ -39,9 +40,6 @@ struct wm8731_priv {
unsigned int sysclk;
};
-#ifdef CONFIG_SPI_MASTER
-static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
-#endif
/*
* wm8731 register cache
@@ -50,60 +48,12 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
* There is no point in caching the reset register
*/
static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
- 0x0097, 0x0097, 0x0079, 0x0079,
- 0x000a, 0x0008, 0x009f, 0x000a,
- 0x0000, 0x0000
+ 0x0097, 0x0097, 0x0079, 0x0079,
+ 0x000a, 0x0008, 0x009f, 0x000a,
+ 0x0000, 0x0000
};
-/*
- * read wm8731 register cache
- */
-static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg == WM8731_RESET)
- return 0;
- if (reg >= WM8731_CACHEREGNUM)
- return -1;
- return cache[reg];
-}
-
-/*
- * write wm8731 register cache
- */
-static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg >= WM8731_CACHEREGNUM)
- return;
- cache[reg] = value;
-}
-
-/*
- * write to the WM8731 register space
- */
-static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8731 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8731_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0)
+#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0)
static const char *wm8731_input_select[] = {"Line In", "Mic"};
static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
@@ -113,20 +63,26 @@ static const struct soc_enum wm8731_enum[] = {
SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
};
+static const DECLARE_TLV_DB_SCALE(in_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
static const struct snd_kcontrol_new wm8731_snd_controls[] = {
-SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
- 0, 127, 0),
+SOC_DOUBLE_R_TLV("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
+ 0, 127, 0, out_tlv),
SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V,
7, 1, 0),
-SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0,
+ in_tlv),
SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
-SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1),
+SOC_SINGLE("Mic Capture Switch", WM8731_APANA, 1, 1, 1),
-SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1),
+SOC_SINGLE_TLV("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1,
+ sidetone_tlv),
SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
@@ -260,12 +216,12 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct wm8731_priv *wm8731 = codec->private_data;
- u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
+ u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3;
int i = get_coeff(wm8731->sysclk, params_rate(params));
u16 srate = (coeff_div[i].sr << 2) |
(coeff_div[i].bosr << 1) | coeff_div[i].usb;
- wm8731_write(codec, WM8731_SRATE, srate);
+ snd_soc_write(codec, WM8731_SRATE, srate);
/* bit size */
switch (params_format(params)) {
@@ -279,7 +235,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
break;
}
- wm8731_write(codec, WM8731_IFACE, iface);
+ snd_soc_write(codec, WM8731_IFACE, iface);
return 0;
}
@@ -291,7 +247,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = socdev->card->codec;
/* set active */
- wm8731_write(codec, WM8731_ACTIVE, 0x0001);
+ snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
return 0;
}
@@ -306,19 +262,19 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream,
/* deactivate */
if (!codec->active) {
udelay(50);
- wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ snd_soc_write(codec, WM8731_ACTIVE, 0x0);
}
}
static int wm8731_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
+ u16 mute_reg = snd_soc_read(codec, WM8731_APDIGI) & 0xfff7;
if (mute)
- wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8);
+ snd_soc_write(codec, WM8731_APDIGI, mute_reg | 0x8);
else
- wm8731_write(codec, WM8731_APDIGI, mute_reg);
+ snd_soc_write(codec, WM8731_APDIGI, mute_reg);
return 0;
}
@@ -396,7 +352,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/* set iface */
- wm8731_write(codec, WM8731_IFACE, iface);
+ snd_soc_write(codec, WM8731_IFACE, iface);
return 0;
}
@@ -412,12 +368,12 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
/* Clear PWROFF, gate CLKOUT, everything else as-is */
- reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
- wm8731_write(codec, WM8731_PWR, reg | 0x0040);
+ reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f;
+ snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
- wm8731_write(codec, WM8731_ACTIVE, 0x0);
- wm8731_write(codec, WM8731_PWR, 0xffff);
+ snd_soc_write(codec, WM8731_ACTIVE, 0x0);
+ snd_soc_write(codec, WM8731_PWR, 0xffff);
break;
}
codec->bias_level = level;
@@ -457,15 +413,17 @@ struct snd_soc_dai wm8731_dai = {
.rates = WM8731_RATES,
.formats = WM8731_FORMATS,},
.ops = &wm8731_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(wm8731_dai);
+#ifdef CONFIG_PM
static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->card->codec;
- wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ snd_soc_write(codec, WM8731_ACTIVE, 0x0);
wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
@@ -488,6 +446,10 @@ static int wm8731_resume(struct platform_device *pdev)
wm8731_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
+#else
+#define wm8731_suspend NULL
+#define wm8731_resume NULL
+#endif
static int wm8731_probe(struct platform_device *pdev)
{
@@ -547,15 +509,16 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
-static int wm8731_register(struct wm8731_priv *wm8731)
+static int wm8731_register(struct wm8731_priv *wm8731,
+ enum snd_soc_control_type control)
{
int ret;
struct snd_soc_codec *codec = &wm8731->codec;
- u16 reg;
if (wm8731_codec) {
dev_err(codec->dev, "Another WM8731 is registered\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto err;
}
mutex_init(&codec->mutex);
@@ -565,8 +528,6 @@ static int wm8731_register(struct wm8731_priv *wm8731)
codec->private_data = wm8731;
codec->name = "WM8731";
codec->owner = THIS_MODULE;
- codec->read = wm8731_read_reg_cache;
- codec->write = wm8731_write;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8731_set_bias_level;
codec->dai = &wm8731_dai;
@@ -576,10 +537,16 @@ static int wm8731_register(struct wm8731_priv *wm8731)
memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg));
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
ret = wm8731_reset(codec);
if (ret < 0) {
- dev_err(codec->dev, "Failed to issue reset\n");
- return ret;
+ dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+ goto err;
}
wm8731_dai.dev = codec->dev;
@@ -587,35 +554,36 @@ static int wm8731_register(struct wm8731_priv *wm8731)
wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch the update bits */
- reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
- wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
- reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
- wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
- reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
- wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
- reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
- wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
+ snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0);
+ snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0);
+ snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0);
+ snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0);
/* Disable bypass path by default */
- reg = wm8731_read_reg_cache(codec, WM8731_APANA);
- wm8731_write(codec, WM8731_APANA, reg & ~0x4);
+ snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0);
wm8731_codec = codec;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
- return ret;
+ goto err;
}
ret = snd_soc_register_dai(&wm8731_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
- return ret;
+ goto err_codec;
}
return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8731);
+ return ret;
}
static void wm8731_unregister(struct wm8731_priv *wm8731)
@@ -628,30 +596,6 @@ static void wm8731_unregister(struct wm8731_priv *wm8731)
}
#if defined(CONFIG_SPI_MASTER)
-static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
-{
- struct spi_transfer t;
- struct spi_message m;
- u8 msg[2];
-
- if (len <= 0)
- return 0;
-
- msg[0] = data[0];
- msg[1] = data[1];
-
- spi_message_init(&m);
- memset(&t, 0, (sizeof t));
-
- t.tx_buf = &msg[0];
- t.len = len;
-
- spi_message_add_tail(&t, &m);
- spi_sync(spi, &m);
-
- return len;
-}
-
static int __devinit wm8731_spi_probe(struct spi_device *spi)
{
struct snd_soc_codec *codec;
@@ -663,12 +607,11 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
codec = &wm8731->codec;
codec->control_data = spi;
- codec->hw_write = (hw_write_t)wm8731_spi_write;
codec->dev = &spi->dev;
dev_set_drvdata(&spi->dev, wm8731);
- return wm8731_register(wm8731);
+ return wm8731_register(wm8731, SND_SOC_SPI);
}
static int __devexit wm8731_spi_remove(struct spi_device *spi)
@@ -680,6 +623,21 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8731_spi_resume(struct spi_device *spi)
+{
+ return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8731_spi_suspend NULL
+#define wm8731_spi_resume NULL
+#endif
+
static struct spi_driver wm8731_spi_driver = {
.driver = {
.name = "wm8731",
@@ -687,6 +645,8 @@ static struct spi_driver wm8731_spi_driver = {
.owner = THIS_MODULE,
},
.probe = wm8731_spi_probe,
+ .suspend = wm8731_spi_suspend,
+ .resume = wm8731_spi_resume,
.remove = __devexit_p(wm8731_spi_remove),
};
#endif /* CONFIG_SPI_MASTER */
@@ -703,14 +663,13 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
codec = &wm8731->codec;
- codec->hw_write = (hw_write_t)i2c_master_send;
i2c_set_clientdata(i2c, wm8731);
codec->control_data = i2c;
codec->dev = &i2c->dev;
- return wm8731_register(wm8731);
+ return wm8731_register(wm8731, SND_SOC_I2C);
}
static __devexit int wm8731_i2c_remove(struct i2c_client *client)
@@ -720,6 +679,21 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&i2c->dev);
+}
+
+static int wm8731_i2c_resume(struct i2c_client *i2c)
+{
+ return snd_soc_resume_device(&i2c->dev);
+}
+#else
+#define wm8731_i2c_suspend NULL
+#define wm8731_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8731_i2c_id[] = {
{ "wm8731", 0 },
{ }
@@ -733,6 +707,8 @@ static struct i2c_driver wm8731_i2c_driver = {
},
.probe = wm8731_i2c_probe,
.remove = __devexit_p(wm8731_i2c_remove),
+ .suspend = wm8731_i2c_suspend,
+ .resume = wm8731_i2c_resume,
.id_table = wm8731_i2c_id,
};
#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index b64509b..4ba1e7e 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -55,50 +55,7 @@ static const u16 wm8750_reg[] = {
0x0079, 0x0079, 0x0079, /* 40 */
};
-/*
- * read wm8750 register cache
- */
-static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg > WM8750_CACHE_REGNUM)
- return -1;
- return cache[reg];
-}
-
-/*
- * write wm8750 register cache
- */
-static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg > WM8750_CACHE_REGNUM)
- return;
- cache[reg] = value;
-}
-
-static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8753 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8750_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0)
+#define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0)
/*
* WM8750 Controls
@@ -594,7 +551,7 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8750_write(codec, WM8750_IFACE, iface);
+ snd_soc_write(codec, WM8750_IFACE, iface);
return 0;
}
@@ -606,8 +563,8 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct wm8750_priv *wm8750 = codec->private_data;
- u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
- u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
+ u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3;
+ u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0;
int coeff = get_coeff(wm8750->sysclk, params_rate(params));
/* bit size */
@@ -626,9 +583,9 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
}
/* set iface & srate */
- wm8750_write(codec, WM8750_IFACE, iface);
+ snd_soc_write(codec, WM8750_IFACE, iface);
if (coeff >= 0)
- wm8750_write(codec, WM8750_SRATE, srate |
+ snd_soc_write(codec, WM8750_SRATE, srate |
(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
return 0;
@@ -637,35 +594,35 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
static int wm8750_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
+ u16 mute_reg = snd_soc_read(codec, WM8750_ADCDAC) & 0xfff7;
if (mute)
- wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
+ snd_soc_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
else
- wm8750_write(codec, WM8750_ADCDAC, mute_reg);
+ snd_soc_write(codec, WM8750_ADCDAC, mute_reg);
return 0;
}
static int wm8750_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
+ u16 pwr_reg = snd_soc_read(codec, WM8750_PWR1) & 0xfe3e;
switch (level) {
case SND_SOC_BIAS_ON:
/* set vmid to 50k and unmute dac */
- wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
+ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
break;
case SND_SOC_BIAS_PREPARE:
/* set vmid to 5k for quick power up */
- wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
+ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
break;
case SND_SOC_BIAS_STANDBY:
/* mute dac and set vmid to 500k, enable VREF */
- wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
+ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
break;
case SND_SOC_BIAS_OFF:
- wm8750_write(codec, WM8750_PWR1, 0x0001);
+ snd_soc_write(codec, WM8750_PWR1, 0x0001);
break;
}
codec->bias_level = level;
@@ -754,15 +711,14 @@ static int wm8750_resume(struct platform_device *pdev)
* initialise the WM8750 driver
* register the mixer and dsp interfaces with the kernel
*/
-static int wm8750_init(struct snd_soc_device *socdev)
+static int wm8750_init(struct snd_soc_device *socdev,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = socdev->card->codec;
int reg, ret = 0;
codec->name = "WM8750";
codec->owner = THIS_MODULE;
- codec->read = wm8750_read_reg_cache;
- codec->write = wm8750_write;
codec->set_bias_level = wm8750_set_bias_level;
codec->dai = &wm8750_dai;
codec->num_dai = 1;
@@ -771,13 +727,23 @@ static int wm8750_init(struct snd_soc_device *socdev)
if (codec->reg_cache == NULL)
return -ENOMEM;
- wm8750_reset(codec);
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ ret = wm8750_reset(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to reset: %d\n", ret);
+ goto err;
+ }
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "wm8750: failed to create pcms\n");
- goto pcm_err;
+ goto err;
}
/* charge output caps */
@@ -786,22 +752,22 @@ static int wm8750_init(struct snd_soc_device *socdev)
schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
/* set the update bits */
- reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
- wm8750_write(codec, WM8750_LDAC, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_RDAC);
- wm8750_write(codec, WM8750_RDAC, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V);
- wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V);
- wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V);
- wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V);
- wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_LINVOL);
- wm8750_write(codec, WM8750_LINVOL, reg | 0x0100);
- reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
- wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_LDAC);
+ snd_soc_write(codec, WM8750_LDAC, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_RDAC);
+ snd_soc_write(codec, WM8750_RDAC, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_LOUT1V);
+ snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_ROUT1V);
+ snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_LOUT2V);
+ snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_ROUT2V);
+ snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_LINVOL);
+ snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8750_RINVOL);
+ snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100);
snd_soc_add_controls(codec, wm8750_snd_controls,
ARRAY_SIZE(wm8750_snd_controls));
@@ -816,7 +782,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-pcm_err:
+err:
kfree(codec->reg_cache);
return ret;
}
@@ -844,7 +810,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c,
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- ret = wm8750_init(socdev);
+ ret = wm8750_init(socdev, SND_SOC_I2C);
if (ret < 0)
pr_err("failed to initialise WM8750\n");
@@ -924,7 +890,7 @@ static int __devinit wm8750_spi_probe(struct spi_device *spi)
codec->control_data = spi;
- ret = wm8750_init(socdev);
+ ret = wm8750_init(socdev, SND_SOC_SPI);
if (ret < 0)
dev_err(&spi->dev, "failed to initialise WM8750\n");
@@ -945,30 +911,6 @@ static struct spi_driver wm8750_spi_driver = {
.probe = wm8750_spi_probe,
.remove = __devexit_p(wm8750_spi_remove),
};
-
-static int wm8750_spi_write(struct spi_device *spi, const char *data, int len)
-{
- struct spi_transfer t;
- struct spi_message m;
- u8 msg[2];
-
- if (len <= 0)
- return 0;
-
- msg[0] = data[0];
- msg[1] = data[1];
-
- spi_message_init(&m);
- memset(&t, 0, (sizeof t));
-
- t.tx_buf = &msg[0];
- t.len = len;
-
- spi_message_add_tail(&t, &m);
- spi_sync(spi, &m);
-
- return len;
-}
#endif
static int wm8750_probe(struct platform_device *pdev)
@@ -1002,13 +944,11 @@ static int wm8750_probe(struct platform_device *pdev)
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8750_add_i2c_device(pdev, setup);
}
#endif
#if defined(CONFIG_SPI_MASTER)
if (setup->spi) {
- codec->hw_write = (hw_write_t)wm8750_spi_write;
ret = spi_register_driver(&wm8750_spi_driver);
if (ret != 0)
printk(KERN_ERR "can't add spi driver");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 49c4b28..d80d414 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1766,6 +1766,21 @@ static int wm8753_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8753_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8753_i2c_suspend NULL
+#define wm8753_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8753_i2c_id[] = {
{ "wm8753", 0 },
{ }
@@ -1779,6 +1794,8 @@ static struct i2c_driver wm8753_i2c_driver = {
},
.probe = wm8753_i2c_probe,
.remove = wm8753_i2c_remove,
+ .suspend = wm8753_i2c_suspend,
+ .resume = wm8753_i2c_resume,
.id_table = wm8753_i2c_id,
};
#endif
@@ -1834,6 +1851,22 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8753_spi_resume(struct spi_device *spi)
+{
+ return snd_soc_resume_device(&spi->dev);
+}
+
+#else
+#define wm8753_spi_suspend NULL
+#define wm8753_spi_resume NULL
+#endif
+
static struct spi_driver wm8753_spi_driver = {
.driver = {
.name = "wm8753",
@@ -1842,6 +1875,8 @@ static struct spi_driver wm8753_spi_driver = {
},
.probe = wm8753_spi_probe,
.remove = __devexit_p(wm8753_spi_remove),
+ .suspend = wm8753_spi_suspend,
+ .resume = wm8753_spi_resume,
};
#endif
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
new file mode 100644
index 0000000..a9829aa
--- /dev/null
+++ b/sound/soc/codecs/wm8776.c
@@ -0,0 +1,744 @@
+/*
+ * wm8776.c -- WM8776 ALSA SoC Audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * TODO: Input ALC/limiter support
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8776.h"
+
+static struct snd_soc_codec *wm8776_codec;
+struct snd_soc_codec_device soc_codec_dev_wm8776;
+
+/* codec private data */
+struct wm8776_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM8776_CACHEREGNUM];
+ int sysclk[2];
+};
+
+#ifdef CONFIG_SPI_MASTER
+static int wm8776_spi_write(struct spi_device *spi, const char *data, int len);
+#endif
+
+static const u16 wm8776_reg[WM8776_CACHEREGNUM] = {
+ 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */
+ 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */
+ 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */
+ 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */
+ 0xa6, 0x01, 0x01
+};
+
+static int wm8776_reset(struct snd_soc_codec *codec)
+{
+ return snd_soc_write(codec, WM8776_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(hp_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -10350, 50, 1);
+
+static const struct snd_kcontrol_new wm8776_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8776_HPLVOL, WM8776_HPRVOL,
+ 0, 127, 0, hp_tlv),
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8776_DACLVOL, WM8776_DACRVOL,
+ 0, 255, 0, dac_tlv),
+SOC_SINGLE("Digital Playback ZC Switch", WM8776_DACCTRL1, 0, 1, 0),
+
+SOC_SINGLE("Deemphasis Switch", WM8776_DACCTRL2, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Volume", WM8776_ADCLVOL, WM8776_ADCRVOL,
+ 0, 255, 0, adc_tlv),
+SOC_DOUBLE("Capture Switch", WM8776_ADCMUX, 7, 6, 1, 1),
+SOC_DOUBLE_R("Capture ZC Switch", WM8776_ADCLVOL, WM8776_ADCRVOL, 8, 1, 0),
+SOC_SINGLE("Capture HPF Switch", WM8776_ADCIFCTRL, 8, 1, 1),
+};
+
+static const struct snd_kcontrol_new inmix_controls[] = {
+SOC_DAPM_SINGLE("AIN1 Switch", WM8776_ADCMUX, 0, 1, 0),
+SOC_DAPM_SINGLE("AIN2 Switch", WM8776_ADCMUX, 1, 1, 0),
+SOC_DAPM_SINGLE("AIN3 Switch", WM8776_ADCMUX, 2, 1, 0),
+SOC_DAPM_SINGLE("AIN4 Switch", WM8776_ADCMUX, 3, 1, 0),
+SOC_DAPM_SINGLE("AIN5 Switch", WM8776_ADCMUX, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new outmix_controls[] = {
+SOC_DAPM_SINGLE("DAC Switch", WM8776_OUTMUX, 0, 1, 0),
+SOC_DAPM_SINGLE("AUX Switch", WM8776_OUTMUX, 1, 1, 0),
+SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("AUX"),
+SND_SOC_DAPM_INPUT("AUX"),
+
+SND_SOC_DAPM_INPUT("AIN1"),
+SND_SOC_DAPM_INPUT("AIN2"),
+SND_SOC_DAPM_INPUT("AIN3"),
+SND_SOC_DAPM_INPUT("AIN4"),
+SND_SOC_DAPM_INPUT("AIN5"),
+
+SND_SOC_DAPM_MIXER("Input Mixer", WM8776_PWRDOWN, 6, 1,
+ inmix_controls, ARRAY_SIZE(inmix_controls)),
+
+SND_SOC_DAPM_ADC("ADC", "Capture", WM8776_PWRDOWN, 1, 1),
+SND_SOC_DAPM_DAC("DAC", "Playback", WM8776_PWRDOWN, 2, 1),
+
+SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+ outmix_controls, ARRAY_SIZE(outmix_controls)),
+
+SND_SOC_DAPM_PGA("Headphone PGA", WM8776_PWRDOWN, 3, 1, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("VOUT"),
+
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+};
+
+static const struct snd_soc_dapm_route routes[] = {
+ { "Input Mixer", "AIN1 Switch", "AIN1" },
+ { "Input Mixer", "AIN2 Switch", "AIN2" },
+ { "Input Mixer", "AIN3 Switch", "AIN3" },
+ { "Input Mixer", "AIN4 Switch", "AIN4" },
+ { "Input Mixer", "AIN5 Switch", "AIN5" },
+
+ { "ADC", NULL, "Input Mixer" },
+
+ { "Output Mixer", "DAC Switch", "DAC" },
+ { "Output Mixer", "AUX Switch", "AUX" },
+ { "Output Mixer", "Bypass Switch", "Input Mixer" },
+
+ { "VOUT", NULL, "Output Mixer" },
+
+ { "Headphone PGA", NULL, "Output Mixer" },
+
+ { "HPOUTL", NULL, "Headphone PGA" },
+ { "HPOUTR", NULL, "Headphone PGA" },
+};
+
+static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int reg, iface, master;
+
+ switch (dai->id) {
+ case WM8776_DAI_DAC:
+ reg = WM8776_DACIFCTRL;
+ master = 0x80;
+ break;
+ case WM8776_DAI_ADC:
+ reg = WM8776_ADCIFCTRL;
+ master = 0x100;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ iface = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ master = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ /* FIXME: CHECK A/B */
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0007;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x00c;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x008;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x004;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Finally, write out the values */
+ snd_soc_update_bits(codec, reg, 0xf, iface);
+ snd_soc_update_bits(codec, WM8776_MSTRCTRL, 0x180, master);
+
+ return 0;
+}
+
+static int mclk_ratios[] = {
+ 128,
+ 192,
+ 256,
+ 384,
+ 512,
+ 768,
+};
+
+static int wm8776_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8776_priv *wm8776 = codec->private_data;
+ int iface_reg, iface;
+ int ratio_shift, master;
+ int i;
+
+ iface = 0;
+
+ switch (dai->id) {
+ case WM8776_DAI_DAC:
+ iface_reg = WM8776_DACIFCTRL;
+ master = 0x80;
+ ratio_shift = 4;
+ break;
+ case WM8776_DAI_ADC:
+ iface_reg = WM8776_ADCIFCTRL;
+ master = 0x100;
+ ratio_shift = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+
+ /* Set word length */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x20;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x30;
+ break;
+ }
+
+ /* Only need to set MCLK/LRCLK ratio if we're master */
+ if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) {
+ for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) {
+ if (wm8776->sysclk[dai->id] / params_rate(params)
+ == mclk_ratios[i])
+ break;
+ }
+
+ if (i == ARRAY_SIZE(mclk_ratios)) {
+ dev_err(codec->dev,
+ "Unable to configure MCLK ratio %d/%d\n",
+ wm8776->sysclk[dai->id], params_rate(params));
+ return -EINVAL;
+ }
+
+ dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]);
+
+ snd_soc_update_bits(codec, WM8776_MSTRCTRL,
+ 0x7 << ratio_shift, i << ratio_shift);
+ } else {
+ dev_dbg(codec->dev, "DAI in slave mode\n");
+ }
+
+ snd_soc_update_bits(codec, iface_reg, 0x30, iface);
+
+ return 0;
+}
+
+static int wm8776_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ return snd_soc_write(codec, WM8776_DACMUTE, !!mute);
+}
+
+static int wm8776_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8776_priv *wm8776 = codec->private_data;
+
+ BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk));
+
+ wm8776->sysclk[dai->id] = freq;
+
+ return 0;
+}
+
+static int wm8776_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Disable the global powerdown; DAPM does the rest */
+ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0);
+ }
+
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 1);
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+
+#define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8776_dac_ops = {
+ .digital_mute = wm8776_mute,
+ .hw_params = wm8776_hw_params,
+ .set_fmt = wm8776_set_fmt,
+ .set_sysclk = wm8776_set_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8776_adc_ops = {
+ .hw_params = wm8776_hw_params,
+ .set_fmt = wm8776_set_fmt,
+ .set_sysclk = wm8776_set_sysclk,
+};
+
+struct snd_soc_dai wm8776_dai[] = {
+ {
+ .name = "WM8776 Playback",
+ .id = WM8776_DAI_DAC,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8776_RATES,
+ .formats = WM8776_FORMATS,
+ },
+ .ops = &wm8776_dac_ops,
+ },
+ {
+ .name = "WM8776 Capture",
+ .id = WM8776_DAI_ADC,
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8776_RATES,
+ .formats = WM8776_FORMATS,
+ },
+ .ops = &wm8776_adc_ops,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8776_dai);
+
+#ifdef CONFIG_PM
+static int wm8776_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8776_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define wm8776_suspend NULL
+#define wm8776_resume NULL
+#endif
+
+static int wm8776_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8776_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8776_codec;
+ codec = wm8776_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8776_snd_controls,
+ ARRAY_SIZE(wm8776_snd_controls));
+ snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets,
+ ARRAY_SIZE(wm8776_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8776_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8776 = {
+ .probe = wm8776_probe,
+ .remove = wm8776_remove,
+ .suspend = wm8776_suspend,
+ .resume = wm8776_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8776);
+
+static int wm8776_register(struct wm8776_priv *wm8776,
+ enum snd_soc_control_type control)
+{
+ int ret, i;
+ struct snd_soc_codec *codec = &wm8776->codec;
+
+ if (wm8776_codec) {
+ dev_err(codec->dev, "Another WM8776 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8776;
+ codec->name = "WM8776";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8776_set_bias_level;
+ codec->dai = wm8776_dai;
+ codec->num_dai = ARRAY_SIZE(wm8776_dai);
+ codec->reg_cache_size = WM8776_CACHEREGNUM;
+ codec->reg_cache = &wm8776->reg_cache;
+
+ memcpy(codec->reg_cache, wm8776_reg, sizeof(wm8776_reg));
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm8776_dai); i++)
+ wm8776_dai[i].dev = codec->dev;
+
+ ret = wm8776_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+ goto err;
+ }
+
+ wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Latch the update bits; right channel only since we always
+ * update both. */
+ snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100);
+
+ wm8776_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai));
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8776);
+ return ret;
+}
+
+static void wm8776_unregister(struct wm8776_priv *wm8776)
+{
+ wm8776_set_bias_level(&wm8776->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai));
+ snd_soc_unregister_codec(&wm8776->codec);
+ kfree(wm8776);
+ wm8776_codec = NULL;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8776_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+
+static int __devinit wm8776_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_codec *codec;
+ struct wm8776_priv *wm8776;
+
+ wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL);
+ if (wm8776 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8776->codec;
+ codec->control_data = spi;
+ codec->hw_write = (hw_write_t)wm8776_spi_write;
+ codec->dev = &spi->dev;
+
+ dev_set_drvdata(&spi->dev, wm8776);
+
+ return wm8776_register(wm8776, SND_SOC_SPI);
+}
+
+static int __devexit wm8776_spi_remove(struct spi_device *spi)
+{
+ struct wm8776_priv *wm8776 = dev_get_drvdata(&spi->dev);
+
+ wm8776_unregister(wm8776);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8776_spi_resume(struct spi_device *spi)
+{
+ return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8776_spi_suspend NULL
+#define wm8776_spi_resume NULL
+#endif
+
+static struct spi_driver wm8776_spi_driver = {
+ .driver = {
+ .name = "wm8776",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8776_spi_probe,
+ .suspend = wm8776_spi_suspend,
+ .resume = wm8776_spi_resume,
+ .remove = __devexit_p(wm8776_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8776_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8776_priv *wm8776;
+ struct snd_soc_codec *codec;
+
+ wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL);
+ if (wm8776 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8776->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8776);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8776_register(wm8776, SND_SOC_I2C);
+}
+
+static __devexit int wm8776_i2c_remove(struct i2c_client *client)
+{
+ struct wm8776_priv *wm8776 = i2c_get_clientdata(client);
+ wm8776_unregister(wm8776);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&i2c->dev);
+}
+
+static int wm8776_i2c_resume(struct i2c_client *i2c)
+{
+ return snd_soc_resume_device(&i2c->dev);
+}
+#else
+#define wm8776_i2c_suspend NULL
+#define wm8776_i2c_resume NULL
+#endif
+
+static const struct i2c_device_id wm8776_i2c_id[] = {
+ { "wm8776", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id);
+
+static struct i2c_driver wm8776_i2c_driver = {
+ .driver = {
+ .name = "wm8776",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8776_i2c_probe,
+ .remove = __devexit_p(wm8776_i2c_remove),
+ .suspend = wm8776_i2c_suspend,
+ .resume = wm8776_i2c_resume,
+ .id_table = wm8776_i2c_id,
+};
+#endif
+
+static int __init wm8776_modinit(void)
+{
+ int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8776_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8776 I2C driver: %d\n",
+ ret);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8776_spi_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8776 SPI driver: %d\n",
+ ret);
+ }
+#endif
+ return 0;
+}
+module_init(wm8776_modinit);
+
+static void __exit wm8776_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8776_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8776_spi_driver);
+#endif
+}
+module_exit(wm8776_exit);
+
+MODULE_DESCRIPTION("ASoC WM8776 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8776.h b/sound/soc/codecs/wm8776.h
new file mode 100644
index 0000000..6606d25
--- /dev/null
+++ b/sound/soc/codecs/wm8776.h
@@ -0,0 +1,51 @@
+/*
+ * wm8776.h -- WM8776 ASoC driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8776_H
+#define _WM8776_H
+
+/* Registers */
+
+#define WM8776_HPLVOL 0x00
+#define WM8776_HPRVOL 0x01
+#define WM8776_HPMASTER 0x02
+#define WM8776_DACLVOL 0x03
+#define WM8776_DACRVOL 0x04
+#define WM8776_DACMASTER 0x05
+#define WM8776_PHASESWAP 0x06
+#define WM8776_DACCTRL1 0x07
+#define WM8776_DACMUTE 0x08
+#define WM8776_DACCTRL2 0x09
+#define WM8776_DACIFCTRL 0x0a
+#define WM8776_ADCIFCTRL 0x0b
+#define WM8776_MSTRCTRL 0x0c
+#define WM8776_PWRDOWN 0x0d
+#define WM8776_ADCLVOL 0x0e
+#define WM8776_ADCRVOL 0x0f
+#define WM8776_ALCCTRL1 0x10
+#define WM8776_ALCCTRL2 0x11
+#define WM8776_ALCCTRL3 0x12
+#define WM8776_NOISEGATE 0x13
+#define WM8776_LIMITER 0x14
+#define WM8776_ADCMUX 0x15
+#define WM8776_OUTMUX 0x16
+#define WM8776_RESET 0x17
+
+#define WM8776_CACHEREGNUM 0x17
+
+#define WM8776_DAI_DAC 0
+#define WM8776_DAI_ADC 1
+
+extern struct snd_soc_dai wm8776_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_wm8776;
+
+#endif
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3c78945..5e9c855 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -116,6 +116,7 @@
#define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c
#define WM8900_REG_DACCTRL_MUTE 0x004
+#define WM8900_REG_DACCTRL_DAC_SB_FILT 0x100
#define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400
#define WM8900_REG_AUDIO3_ADCLRC_DIR 0x0800
@@ -182,111 +183,20 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = {
/* Remaining registers all zero */
};
-/*
- * read wm8900 register cache
- */
-static inline unsigned int wm8900_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
-
- BUG_ON(reg >= WM8900_MAXREG);
-
- if (reg == WM8900_REG_ID)
- return 0;
-
- return cache[reg];
-}
-
-/*
- * write wm8900 register cache
- */
-static inline void wm8900_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
-
- BUG_ON(reg >= WM8900_MAXREG);
-
- cache[reg] = value;
-}
-
-/*
- * write to the WM8900 register space
- */
-static int wm8900_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- if (value == wm8900_read_reg_cache(codec, reg))
- return 0;
-
- /* data is
- * D15..D9 WM8900 register offset
- * D8...D0 register data
- */
- data[0] = reg;
- data[1] = value >> 8;
- data[2] = value & 0x00ff;
-
- wm8900_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 3) == 3)
- return 0;
- else
- return -EIO;
-}
-
-/*
- * Read from the wm8900.
- */
-static unsigned int wm8900_chip_read(struct snd_soc_codec *codec, u8 reg)
-{
- struct i2c_msg xfer[2];
- u16 data;
- int ret;
- struct i2c_client *client = codec->control_data;
-
- BUG_ON(reg != WM8900_REG_ID && reg != WM8900_REG_POWER1);
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = 1;
- xfer[0].buf = &reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = 2;
- xfer[1].buf = (u8 *)&data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret != 2) {
- printk(KERN_CRIT "i2c_transfer returned %d\n", ret);
- return 0;
- }
-
- return (data >> 8) | ((data & 0xff) << 8);
-}
-
-/*
- * Read from the WM8900 register space. Most registers can't be read
- * and are therefore supplied from cache.
- */
-static unsigned int wm8900_read(struct snd_soc_codec *codec, unsigned int reg)
+static int wm8900_volatile_register(unsigned int reg)
{
switch (reg) {
case WM8900_REG_ID:
- return wm8900_chip_read(codec, reg);
+ case WM8900_REG_POWER1:
+ return 1;
default:
- return wm8900_read_reg_cache(codec, reg);
+ return 0;
}
}
static void wm8900_reset(struct snd_soc_codec *codec)
{
- wm8900_write(codec, WM8900_REG_RESET, 0);
+ snd_soc_write(codec, WM8900_REG_RESET, 0);
memcpy(codec->reg_cache, wm8900_reg_defaults,
sizeof(codec->reg_cache));
@@ -296,14 +206,14 @@ static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- u16 hpctl1 = wm8900_read(codec, WM8900_REG_HPCTL1);
+ u16 hpctl1 = snd_soc_read(codec, WM8900_REG_HPCTL1);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* Clamp headphone outputs */
hpctl1 = WM8900_REG_HPCTL1_HP_CLAMP_IP |
WM8900_REG_HPCTL1_HP_CLAMP_OP;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
break;
case SND_SOC_DAPM_POST_PMU:
@@ -312,41 +222,41 @@ static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT |
WM8900_REG_HPCTL1_HP_SHORT2 |
WM8900_REG_HPCTL1_HP_IPSTAGE_ENA;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
msleep(400);
/* Enable the output stage */
hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_OP;
hpctl1 |= WM8900_REG_HPCTL1_HP_OPSTAGE_ENA;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
/* Remove the shorts */
hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT2;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
break;
case SND_SOC_DAPM_PRE_PMD:
/* Short the output */
hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
/* Disable the output stage */
hpctl1 &= ~WM8900_REG_HPCTL1_HP_OPSTAGE_ENA;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
/* Clamp the outputs and power down input */
hpctl1 |= WM8900_REG_HPCTL1_HP_CLAMP_IP |
WM8900_REG_HPCTL1_HP_CLAMP_OP;
hpctl1 &= ~WM8900_REG_HPCTL1_HP_IPSTAGE_ENA;
- wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1);
break;
case SND_SOC_DAPM_POST_PMD:
/* Disable everything */
- wm8900_write(codec, WM8900_REG_HPCTL1, 0);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, 0);
break;
default:
@@ -439,7 +349,6 @@ SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1),
SOC_ENUM("DAC Mute Rate", dac_mute_rate),
SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0),
SOC_ENUM("DAC Deemphasis", dac_deemphasis),
-SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0),
SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL,
12, 1, 0),
@@ -723,7 +632,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = socdev->card->codec;
u16 reg;
- reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60;
+ reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
@@ -741,7 +650,18 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- wm8900_write(codec, WM8900_REG_AUDIO1, reg);
+ snd_soc_write(codec, WM8900_REG_AUDIO1, reg);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ reg = snd_soc_read(codec, WM8900_REG_DACCTRL);
+
+ if (params_rate(params) <= 24000)
+ reg |= WM8900_REG_DACCTRL_DAC_SB_FILT;
+ else
+ reg &= ~WM8900_REG_DACCTRL_DAC_SB_FILT;
+
+ snd_soc_write(codec, WM8900_REG_DACCTRL, reg);
+ }
return 0;
}
@@ -834,18 +754,18 @@ static int wm8900_set_fll(struct snd_soc_codec *codec,
return 0;
/* The digital side should be disabled during any change. */
- reg = wm8900_read(codec, WM8900_REG_POWER1);
- wm8900_write(codec, WM8900_REG_POWER1,
+ reg = snd_soc_read(codec, WM8900_REG_POWER1);
+ snd_soc_write(codec, WM8900_REG_POWER1,
reg & (~WM8900_REG_POWER1_FLL_ENA));
/* Disable the FLL? */
if (!freq_in || !freq_out) {
- reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
- wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+ snd_soc_write(codec, WM8900_REG_CLOCKING1,
reg & (~WM8900_REG_CLOCKING1_MCLK_SRC));
- reg = wm8900_read(codec, WM8900_REG_FLLCTL1);
- wm8900_write(codec, WM8900_REG_FLLCTL1,
+ reg = snd_soc_read(codec, WM8900_REG_FLLCTL1);
+ snd_soc_write(codec, WM8900_REG_FLLCTL1,
reg & (~WM8900_REG_FLLCTL1_OSC_ENA));
wm8900->fll_in = freq_in;
@@ -862,33 +782,33 @@ static int wm8900_set_fll(struct snd_soc_codec *codec,
/* The osclilator *MUST* be enabled before we enable the
* digital circuit. */
- wm8900_write(codec, WM8900_REG_FLLCTL1,
+ snd_soc_write(codec, WM8900_REG_FLLCTL1,
fll_div.fll_ratio | WM8900_REG_FLLCTL1_OSC_ENA);
- wm8900_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5);
- wm8900_write(codec, WM8900_REG_FLLCTL5,
+ snd_soc_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5);
+ snd_soc_write(codec, WM8900_REG_FLLCTL5,
(fll_div.fllclk_div << 6) | (fll_div.n & 0x1f));
if (fll_div.k) {
- wm8900_write(codec, WM8900_REG_FLLCTL2,
+ snd_soc_write(codec, WM8900_REG_FLLCTL2,
(fll_div.k >> 8) | 0x100);
- wm8900_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff);
+ snd_soc_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff);
} else
- wm8900_write(codec, WM8900_REG_FLLCTL2, 0);
+ snd_soc_write(codec, WM8900_REG_FLLCTL2, 0);
if (fll_div.fll_slow_lock_ref)
- wm8900_write(codec, WM8900_REG_FLLCTL6,
+ snd_soc_write(codec, WM8900_REG_FLLCTL6,
WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF);
else
- wm8900_write(codec, WM8900_REG_FLLCTL6, 0);
+ snd_soc_write(codec, WM8900_REG_FLLCTL6, 0);
- reg = wm8900_read(codec, WM8900_REG_POWER1);
- wm8900_write(codec, WM8900_REG_POWER1,
+ reg = snd_soc_read(codec, WM8900_REG_POWER1);
+ snd_soc_write(codec, WM8900_REG_POWER1,
reg | WM8900_REG_POWER1_FLL_ENA);
reenable:
- reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
- wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+ snd_soc_write(codec, WM8900_REG_CLOCKING1,
reg | WM8900_REG_CLOCKING1_MCLK_SRC);
return 0;
@@ -908,38 +828,38 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8900_BCLK_DIV:
- reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
- wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+ snd_soc_write(codec, WM8900_REG_CLOCKING1,
div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK));
break;
case WM8900_OPCLK_DIV:
- reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
- wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+ snd_soc_write(codec, WM8900_REG_CLOCKING1,
div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK));
break;
case WM8900_DAC_LRCLK:
- reg = wm8900_read(codec, WM8900_REG_AUDIO4);
- wm8900_write(codec, WM8900_REG_AUDIO4,
+ reg = snd_soc_read(codec, WM8900_REG_AUDIO4);
+ snd_soc_write(codec, WM8900_REG_AUDIO4,
div | (reg & WM8900_LRC_MASK));
break;
case WM8900_ADC_LRCLK:
- reg = wm8900_read(codec, WM8900_REG_AUDIO3);
- wm8900_write(codec, WM8900_REG_AUDIO3,
+ reg = snd_soc_read(codec, WM8900_REG_AUDIO3);
+ snd_soc_write(codec, WM8900_REG_AUDIO3,
div | (reg & WM8900_LRC_MASK));
break;
case WM8900_DAC_CLKDIV:
- reg = wm8900_read(codec, WM8900_REG_CLOCKING2);
- wm8900_write(codec, WM8900_REG_CLOCKING2,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING2);
+ snd_soc_write(codec, WM8900_REG_CLOCKING2,
div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV));
break;
case WM8900_ADC_CLKDIV:
- reg = wm8900_read(codec, WM8900_REG_CLOCKING2);
- wm8900_write(codec, WM8900_REG_CLOCKING2,
+ reg = snd_soc_read(codec, WM8900_REG_CLOCKING2);
+ snd_soc_write(codec, WM8900_REG_CLOCKING2,
div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV));
break;
case WM8900_LRCLK_MODE:
- reg = wm8900_read(codec, WM8900_REG_DACCTRL);
- wm8900_write(codec, WM8900_REG_DACCTRL,
+ reg = snd_soc_read(codec, WM8900_REG_DACCTRL);
+ snd_soc_write(codec, WM8900_REG_DACCTRL,
div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE));
break;
default:
@@ -956,10 +876,10 @@ static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
unsigned int clocking1, aif1, aif3, aif4;
- clocking1 = wm8900_read(codec, WM8900_REG_CLOCKING1);
- aif1 = wm8900_read(codec, WM8900_REG_AUDIO1);
- aif3 = wm8900_read(codec, WM8900_REG_AUDIO3);
- aif4 = wm8900_read(codec, WM8900_REG_AUDIO4);
+ clocking1 = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+ aif1 = snd_soc_read(codec, WM8900_REG_AUDIO1);
+ aif3 = snd_soc_read(codec, WM8900_REG_AUDIO3);
+ aif4 = snd_soc_read(codec, WM8900_REG_AUDIO4);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1055,10 +975,10 @@ static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8900_write(codec, WM8900_REG_CLOCKING1, clocking1);
- wm8900_write(codec, WM8900_REG_AUDIO1, aif1);
- wm8900_write(codec, WM8900_REG_AUDIO3, aif3);
- wm8900_write(codec, WM8900_REG_AUDIO4, aif4);
+ snd_soc_write(codec, WM8900_REG_CLOCKING1, clocking1);
+ snd_soc_write(codec, WM8900_REG_AUDIO1, aif1);
+ snd_soc_write(codec, WM8900_REG_AUDIO3, aif3);
+ snd_soc_write(codec, WM8900_REG_AUDIO4, aif4);
return 0;
}
@@ -1068,14 +988,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
- reg = wm8900_read(codec, WM8900_REG_DACCTRL);
+ reg = snd_soc_read(codec, WM8900_REG_DACCTRL);
if (mute)
reg |= WM8900_REG_DACCTRL_MUTE;
else
reg &= ~WM8900_REG_DACCTRL_MUTE;
- wm8900_write(codec, WM8900_REG_DACCTRL, reg);
+ snd_soc_write(codec, WM8900_REG_DACCTRL, reg);
return 0;
}
@@ -1124,11 +1044,11 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
/* Enable thermal shutdown */
- reg = wm8900_read(codec, WM8900_REG_GPIO);
- wm8900_write(codec, WM8900_REG_GPIO,
+ reg = snd_soc_read(codec, WM8900_REG_GPIO);
+ snd_soc_write(codec, WM8900_REG_GPIO,
reg | WM8900_REG_GPIO_TEMP_ENA);
- reg = wm8900_read(codec, WM8900_REG_ADDCTL);
- wm8900_write(codec, WM8900_REG_ADDCTL,
+ reg = snd_soc_read(codec, WM8900_REG_ADDCTL);
+ snd_soc_write(codec, WM8900_REG_ADDCTL,
reg | WM8900_REG_ADDCTL_TEMP_SD);
break;
@@ -1139,69 +1059,69 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
/* Charge capacitors if initial power up */
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* STARTUP_BIAS_ENA on */
- wm8900_write(codec, WM8900_REG_POWER1,
+ snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_STARTUP_BIAS_ENA);
/* Startup bias mode */
- wm8900_write(codec, WM8900_REG_ADDCTL,
+ snd_soc_write(codec, WM8900_REG_ADDCTL,
WM8900_REG_ADDCTL_BIAS_SRC |
WM8900_REG_ADDCTL_VMID_SOFTST);
/* VMID 2x50k */
- wm8900_write(codec, WM8900_REG_POWER1,
+ snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_STARTUP_BIAS_ENA | 0x1);
/* Allow capacitors to charge */
schedule_timeout_interruptible(msecs_to_jiffies(400));
/* Enable bias */
- wm8900_write(codec, WM8900_REG_POWER1,
+ snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_STARTUP_BIAS_ENA |
WM8900_REG_POWER1_BIAS_ENA | 0x1);
- wm8900_write(codec, WM8900_REG_ADDCTL, 0);
+ snd_soc_write(codec, WM8900_REG_ADDCTL, 0);
- wm8900_write(codec, WM8900_REG_POWER1,
+ snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_BIAS_ENA | 0x1);
}
- reg = wm8900_read(codec, WM8900_REG_POWER1);
- wm8900_write(codec, WM8900_REG_POWER1,
+ reg = snd_soc_read(codec, WM8900_REG_POWER1);
+ snd_soc_write(codec, WM8900_REG_POWER1,
(reg & WM8900_REG_POWER1_FLL_ENA) |
WM8900_REG_POWER1_BIAS_ENA | 0x1);
- wm8900_write(codec, WM8900_REG_POWER2,
+ snd_soc_write(codec, WM8900_REG_POWER2,
WM8900_REG_POWER2_SYSCLK_ENA);
- wm8900_write(codec, WM8900_REG_POWER3, 0);
+ snd_soc_write(codec, WM8900_REG_POWER3, 0);
break;
case SND_SOC_BIAS_OFF:
/* Startup bias enable */
- reg = wm8900_read(codec, WM8900_REG_POWER1);
- wm8900_write(codec, WM8900_REG_POWER1,
+ reg = snd_soc_read(codec, WM8900_REG_POWER1);
+ snd_soc_write(codec, WM8900_REG_POWER1,
reg & WM8900_REG_POWER1_STARTUP_BIAS_ENA);
- wm8900_write(codec, WM8900_REG_ADDCTL,
+ snd_soc_write(codec, WM8900_REG_ADDCTL,
WM8900_REG_ADDCTL_BIAS_SRC |
WM8900_REG_ADDCTL_VMID_SOFTST);
/* Discharge caps */
- wm8900_write(codec, WM8900_REG_POWER1,
+ snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_STARTUP_BIAS_ENA);
schedule_timeout_interruptible(msecs_to_jiffies(500));
/* Remove clamp */
- wm8900_write(codec, WM8900_REG_HPCTL1, 0);
+ snd_soc_write(codec, WM8900_REG_HPCTL1, 0);
/* Power down */
- wm8900_write(codec, WM8900_REG_ADDCTL, 0);
- wm8900_write(codec, WM8900_REG_POWER1, 0);
- wm8900_write(codec, WM8900_REG_POWER2, 0);
- wm8900_write(codec, WM8900_REG_POWER3, 0);
+ snd_soc_write(codec, WM8900_REG_ADDCTL, 0);
+ snd_soc_write(codec, WM8900_REG_POWER1, 0);
+ snd_soc_write(codec, WM8900_REG_POWER2, 0);
+ snd_soc_write(codec, WM8900_REG_POWER3, 0);
/* Need to let things settle before stopping the clock
* to ensure that restart works, see "Stopping the
* master clock" in the datasheet. */
schedule_timeout_interruptible(msecs_to_jiffies(1));
- wm8900_write(codec, WM8900_REG_POWER2,
+ snd_soc_write(codec, WM8900_REG_POWER2,
WM8900_REG_POWER2_SYSCLK_ENA);
break;
}
@@ -1264,7 +1184,7 @@ static int wm8900_resume(struct platform_device *pdev)
if (cache) {
for (i = 0; i < WM8900_MAXREG; i++)
- wm8900_write(codec, i, cache[i]);
+ snd_soc_write(codec, i, cache[i]);
kfree(cache);
} else
dev_err(&pdev->dev, "Unable to allocate register cache\n");
@@ -1297,16 +1217,20 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
codec->name = "WM8900";
codec->owner = THIS_MODULE;
- codec->read = wm8900_read;
- codec->write = wm8900_write;
codec->dai = &wm8900_dai;
codec->num_dai = 1;
- codec->hw_write = (hw_write_t)i2c_master_send;
codec->control_data = i2c;
codec->set_bias_level = wm8900_set_bias_level;
+ codec->volatile_register = wm8900_volatile_register;
codec->dev = &i2c->dev;
- reg = wm8900_read(codec, WM8900_REG_ID);
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ reg = snd_soc_read(codec, WM8900_REG_ID);
if (reg != 0x8900) {
dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg);
ret = -ENODEV;
@@ -1314,7 +1238,7 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
}
/* Read back from the chip */
- reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
+ reg = snd_soc_read(codec, WM8900_REG_POWER1);
reg = (reg >> 12) & 0xf;
dev_info(&i2c->dev, "WM8900 revision %d\n", reg);
@@ -1324,29 +1248,29 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch the volume update bits */
- wm8900_write(codec, WM8900_REG_LINVOL,
- wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
- wm8900_write(codec, WM8900_REG_RINVOL,
- wm8900_read(codec, WM8900_REG_RINVOL) | 0x100);
- wm8900_write(codec, WM8900_REG_LOUT1CTL,
- wm8900_read(codec, WM8900_REG_LOUT1CTL) | 0x100);
- wm8900_write(codec, WM8900_REG_ROUT1CTL,
- wm8900_read(codec, WM8900_REG_ROUT1CTL) | 0x100);
- wm8900_write(codec, WM8900_REG_LOUT2CTL,
- wm8900_read(codec, WM8900_REG_LOUT2CTL) | 0x100);
- wm8900_write(codec, WM8900_REG_ROUT2CTL,
- wm8900_read(codec, WM8900_REG_ROUT2CTL) | 0x100);
- wm8900_write(codec, WM8900_REG_LDAC_DV,
- wm8900_read(codec, WM8900_REG_LDAC_DV) | 0x100);
- wm8900_write(codec, WM8900_REG_RDAC_DV,
- wm8900_read(codec, WM8900_REG_RDAC_DV) | 0x100);
- wm8900_write(codec, WM8900_REG_LADC_DV,
- wm8900_read(codec, WM8900_REG_LADC_DV) | 0x100);
- wm8900_write(codec, WM8900_REG_RADC_DV,
- wm8900_read(codec, WM8900_REG_RADC_DV) | 0x100);
+ snd_soc_write(codec, WM8900_REG_LINVOL,
+ snd_soc_read(codec, WM8900_REG_LINVOL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_RINVOL,
+ snd_soc_read(codec, WM8900_REG_RINVOL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_LOUT1CTL,
+ snd_soc_read(codec, WM8900_REG_LOUT1CTL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_ROUT1CTL,
+ snd_soc_read(codec, WM8900_REG_ROUT1CTL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_LOUT2CTL,
+ snd_soc_read(codec, WM8900_REG_LOUT2CTL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_ROUT2CTL,
+ snd_soc_read(codec, WM8900_REG_ROUT2CTL) | 0x100);
+ snd_soc_write(codec, WM8900_REG_LDAC_DV,
+ snd_soc_read(codec, WM8900_REG_LDAC_DV) | 0x100);
+ snd_soc_write(codec, WM8900_REG_RDAC_DV,
+ snd_soc_read(codec, WM8900_REG_RDAC_DV) | 0x100);
+ snd_soc_write(codec, WM8900_REG_LADC_DV,
+ snd_soc_read(codec, WM8900_REG_LADC_DV) | 0x100);
+ snd_soc_write(codec, WM8900_REG_RADC_DV,
+ snd_soc_read(codec, WM8900_REG_RADC_DV) | 0x100);
/* Set the DAC and mixer output bias */
- wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
+ snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
wm8900_dai.dev = &i2c->dev;
@@ -1388,6 +1312,21 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8900_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8900_i2c_suspend NULL
+#define wm8900_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8900_i2c_id[] = {
{ "wm8900", 0 },
{ }
@@ -1401,6 +1340,8 @@ static struct i2c_driver wm8900_i2c_driver = {
},
.probe = wm8900_i2c_probe,
.remove = __devexit_p(wm8900_i2c_remove),
+ .suspend = wm8900_i2c_suspend,
+ .resume = wm8900_i2c_resume,
.id_table = wm8900_i2c_id,
};
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index e8d2e3e..fe1307b 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -225,94 +225,18 @@ struct wm8903_priv {
struct snd_pcm_substream *slave_substream;
};
-
-static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
-
- BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults));
-
- return cache[reg];
-}
-
-static unsigned int wm8903_hw_read(struct snd_soc_codec *codec, u8 reg)
-{
- struct i2c_msg xfer[2];
- u16 data;
- int ret;
- struct i2c_client *client = codec->control_data;
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = 1;
- xfer[0].buf = &reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = 2;
- xfer[1].buf = (u8 *)&data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret != 2) {
- pr_err("i2c_transfer returned %d\n", ret);
- return 0;
- }
-
- return (data >> 8) | ((data & 0xff) << 8);
-}
-
-static unsigned int wm8903_read(struct snd_soc_codec *codec,
- unsigned int reg)
+static int wm8903_volatile_register(unsigned int reg)
{
switch (reg) {
case WM8903_SW_RESET_AND_ID:
case WM8903_REVISION_NUMBER:
case WM8903_INTERRUPT_STATUS_1:
case WM8903_WRITE_SEQUENCER_4:
- return wm8903_hw_read(codec, reg);
+ return 1;
default:
- return wm8903_read_reg_cache(codec, reg);
- }
-}
-
-static void wm8903_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
-
- BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults));
-
- switch (reg) {
- case WM8903_SW_RESET_AND_ID:
- case WM8903_REVISION_NUMBER:
- break;
-
- default:
- cache[reg] = value;
- break;
- }
-}
-
-static int wm8903_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- wm8903_write_reg_cache(codec, reg, value);
-
- /* Data format is 1 byte of address followed by 2 bytes of data */
- data[0] = reg;
- data[1] = (value >> 8) & 0xff;
- data[2] = value & 0xff;
-
- if (codec->hw_write(codec->control_data, data, 3) == 2)
return 0;
- else
- return -EIO;
+ }
}
static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
@@ -323,13 +247,13 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
BUG_ON(start > 48);
/* Enable the sequencer */
- reg[0] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_0);
+ reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0);
reg[0] |= WM8903_WSEQ_ENA;
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
dev_dbg(&i2c->dev, "Starting sequence at %d\n", start);
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_3,
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3,
start | WM8903_WSEQ_START);
/* Wait for it to complete. If we have the interrupt wired up then
@@ -339,13 +263,13 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
do {
msleep(10);
- reg[4] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_4);
+ reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4);
} while (reg[4] & WM8903_WSEQ_BUSY);
dev_dbg(&i2c->dev, "Sequence complete\n");
/* Disable the sequencer again */
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_0,
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
reg[0] & ~WM8903_WSEQ_ENA);
return 0;
@@ -357,12 +281,12 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
/* There really ought to be something better we can do here :/ */
for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
- cache[i] = wm8903_hw_read(codec, i);
+ cache[i] = codec->hw_read(codec, i);
}
static void wm8903_reset(struct snd_soc_codec *codec)
{
- wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+ snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0);
memcpy(codec->reg_cache, wm8903_reg_defaults,
sizeof(wm8903_reg_defaults));
}
@@ -423,52 +347,52 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
}
if (event & SND_SOC_DAPM_PRE_PMU) {
- val = wm8903_read(codec, reg);
+ val = snd_soc_read(codec, reg);
/* Short the output */
val &= ~(WM8903_OUTPUT_SHORT << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
}
if (event & SND_SOC_DAPM_POST_PMU) {
- val = wm8903_read(codec, reg);
+ val = snd_soc_read(codec, reg);
val |= (WM8903_OUTPUT_IN << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
val |= (WM8903_OUTPUT_INT << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
/* Turn on the output ENA_OUTP */
val |= (WM8903_OUTPUT_OUT << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
/* Enable the DC servo */
- dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
dcs_reg |= dcs_bit;
- wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+ snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
/* Remove the short */
val |= (WM8903_OUTPUT_SHORT << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
}
if (event & SND_SOC_DAPM_PRE_PMD) {
- val = wm8903_read(codec, reg);
+ val = snd_soc_read(codec, reg);
/* Short the output */
val &= ~(WM8903_OUTPUT_SHORT << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
/* Disable the DC servo */
- dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
dcs_reg &= ~dcs_bit;
- wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+ snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
/* Then disable the intermediate and output stages */
val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
WM8903_OUTPUT_IN) << shift);
- wm8903_write(codec, reg, val);
+ snd_soc_write(codec, reg, val);
}
return 0;
@@ -492,13 +416,13 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
u16 reg;
int ret;
- reg = wm8903_read(codec, WM8903_CLASS_W_0);
+ reg = snd_soc_read(codec, WM8903_CLASS_W_0);
/* Turn it off if we're about to enable bypass */
if (ucontrol->value.integer.value[0]) {
if (wm8903->class_w_users == 0) {
dev_dbg(&i2c->dev, "Disabling Class W\n");
- wm8903_write(codec, WM8903_CLASS_W_0, reg &
+ snd_soc_write(codec, WM8903_CLASS_W_0, reg &
~(WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V));
}
wm8903->class_w_users++;
@@ -511,7 +435,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
if (!ucontrol->value.integer.value[0]) {
if (wm8903->class_w_users == 1) {
dev_dbg(&i2c->dev, "Enabling Class W\n");
- wm8903_write(codec, WM8903_CLASS_W_0, reg |
+ snd_soc_write(codec, WM8903_CLASS_W_0, reg |
WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V);
}
wm8903->class_w_users--;
@@ -715,8 +639,6 @@ SOC_ENUM("DAC Soft Mute Rate", soft_mute),
SOC_ENUM("DAC Mute Mode", mute_mode),
SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0),
SOC_ENUM("DAC De-emphasis", dac_deemphasis),
-SOC_SINGLE("DAC Sloping Stopband Filter Switch",
- WM8903_DAC_DIGITAL_1, 11, 1, 0),
SOC_ENUM("DAC Companding Mode", dac_companding),
SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0),
@@ -1011,55 +933,55 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
- reg = wm8903_read(codec, WM8903_VMID_CONTROL_0);
+ reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
reg &= ~(WM8903_VMID_RES_MASK);
reg |= WM8903_VMID_RES_50K;
- wm8903_write(codec, WM8903_VMID_CONTROL_0, reg);
+ snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
- wm8903_write(codec, WM8903_CLOCK_RATES_2,
+ snd_soc_write(codec, WM8903_CLOCK_RATES_2,
WM8903_CLK_SYS_ENA);
/* Change DC servo dither level in startup sequence */
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
- wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
+ snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
/* Enable low impedence charge pump output */
- reg = wm8903_read(codec,
+ reg = snd_soc_read(codec,
WM8903_CONTROL_INTERFACE_TEST_1);
- wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
+ snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
reg | WM8903_TEST_KEY);
- reg2 = wm8903_read(codec, WM8903_CHARGE_PUMP_TEST_1);
- wm8903_write(codec, WM8903_CHARGE_PUMP_TEST_1,
+ reg2 = snd_soc_read(codec, WM8903_CHARGE_PUMP_TEST_1);
+ snd_soc_write(codec, WM8903_CHARGE_PUMP_TEST_1,
reg2 | WM8903_CP_SW_KELVIN_MODE_MASK);
- wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
+ snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
reg);
/* By default no bypass paths are enabled so
* enable Class W support.
*/
dev_dbg(&i2c->dev, "Enabling Class W\n");
- wm8903_write(codec, WM8903_CLASS_W_0, reg |
+ snd_soc_write(codec, WM8903_CLASS_W_0, reg |
WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V);
}
- reg = wm8903_read(codec, WM8903_VMID_CONTROL_0);
+ reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
reg &= ~(WM8903_VMID_RES_MASK);
reg |= WM8903_VMID_RES_250K;
- wm8903_write(codec, WM8903_VMID_CONTROL_0, reg);
+ snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
break;
case SND_SOC_BIAS_OFF:
wm8903_run_sequence(codec, 32);
- reg = wm8903_read(codec, WM8903_CLOCK_RATES_2);
+ reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2);
reg &= ~WM8903_CLK_SYS_ENA;
- wm8903_write(codec, WM8903_CLOCK_RATES_2, reg);
+ snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg);
break;
}
@@ -1083,7 +1005,7 @@ static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1);
+ u16 aif1 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_1);
aif1 &= ~(WM8903_LRCLK_DIR | WM8903_BCLK_DIR | WM8903_AIF_FMT_MASK |
WM8903_AIF_LRCLK_INV | WM8903_AIF_BCLK_INV);
@@ -1161,7 +1083,7 @@ static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
+ snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
return 0;
}
@@ -1171,14 +1093,14 @@ static int wm8903_digital_mute(struct snd_soc_dai *codec_dai, int mute)
struct snd_soc_codec *codec = codec_dai->codec;
u16 reg;
- reg = wm8903_read(codec, WM8903_DAC_DIGITAL_1);
+ reg = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
if (mute)
reg |= WM8903_DAC_MUTE;
else
reg &= ~WM8903_DAC_MUTE;
- wm8903_write(codec, WM8903_DAC_DIGITAL_1, reg);
+ snd_soc_write(codec, WM8903_DAC_DIGITAL_1, reg);
return 0;
}
@@ -1368,17 +1290,24 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
int cur_val;
int clk_sys;
- u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1);
- u16 aif2 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_2);
- u16 aif3 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_3);
- u16 clock0 = wm8903_read(codec, WM8903_CLOCK_RATES_0);
- u16 clock1 = wm8903_read(codec, WM8903_CLOCK_RATES_1);
+ u16 aif1 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_1);
+ u16 aif2 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_2);
+ u16 aif3 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_3);
+ u16 clock0 = snd_soc_read(codec, WM8903_CLOCK_RATES_0);
+ u16 clock1 = snd_soc_read(codec, WM8903_CLOCK_RATES_1);
+ u16 dac_digital1 = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
if (substream == wm8903->slave_substream) {
dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
return 0;
}
+ /* Enable sloping stopband filter for low sample rates */
+ if (fs <= 24000)
+ dac_digital1 |= WM8903_DAC_SB_FILT;
+ else
+ dac_digital1 &= ~WM8903_DAC_SB_FILT;
+
/* Configure sample rate logic for DSP - choose nearest rate */
dsp_config = 0;
best_val = abs(sample_rates[dsp_config].rate - fs);
@@ -1498,11 +1427,12 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
aif2 |= bclk_divs[bclk_div].div;
aif3 |= bclk / fs;
- wm8903_write(codec, WM8903_CLOCK_RATES_0, clock0);
- wm8903_write(codec, WM8903_CLOCK_RATES_1, clock1);
- wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
- wm8903_write(codec, WM8903_AUDIO_INTERFACE_2, aif2);
- wm8903_write(codec, WM8903_AUDIO_INTERFACE_3, aif3);
+ snd_soc_write(codec, WM8903_CLOCK_RATES_0, clock0);
+ snd_soc_write(codec, WM8903_CLOCK_RATES_1, clock1);
+ snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
+ snd_soc_write(codec, WM8903_AUDIO_INTERFACE_2, aif2);
+ snd_soc_write(codec, WM8903_AUDIO_INTERFACE_3, aif3);
+ snd_soc_write(codec, WM8903_DAC_DIGITAL_1, dac_digital1);
return 0;
}
@@ -1587,7 +1517,7 @@ static int wm8903_resume(struct platform_device *pdev)
if (tmp_cache) {
for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
if (tmp_cache[i] != reg_cache[i])
- wm8903_write(codec, i, tmp_cache[i]);
+ snd_soc_write(codec, i, tmp_cache[i]);
} else {
dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
}
@@ -1618,9 +1548,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
codec->dev = &i2c->dev;
codec->name = "WM8903";
codec->owner = THIS_MODULE;
- codec->read = wm8903_read;
- codec->write = wm8903_write;
- codec->hw_write = (hw_write_t)i2c_master_send;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8903_set_bias_level;
codec->dai = &wm8903_dai;
@@ -1628,18 +1555,25 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
codec->reg_cache = &wm8903->reg_cache[0];
codec->private_data = wm8903;
+ codec->volatile_register = wm8903_volatile_register;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
- val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ if (ret != 0) {
+ dev_err(&i2c->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ val = snd_soc_read(codec, WM8903_SW_RESET_AND_ID);
if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
dev_err(&i2c->dev,
"Device with ID register %x is not a WM8903\n", val);
return -ENODEV;
}
- val = wm8903_read(codec, WM8903_REVISION_NUMBER);
+ val = snd_soc_read(codec, WM8903_REVISION_NUMBER);
dev_info(&i2c->dev, "WM8903 revision %d\n",
val & WM8903_CHIP_REV_MASK);
@@ -1649,35 +1583,35 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch volume update bits */
- val = wm8903_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
+ val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
val |= WM8903_ADCVU;
- wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
- wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
+ snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
+ snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
- val = wm8903_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
+ val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
val |= WM8903_DACVU;
- wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
- wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
+ snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
+ snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
- val = wm8903_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
+ val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
val |= WM8903_HPOUTVU;
- wm8903_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
- wm8903_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
- val = wm8903_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
+ val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
val |= WM8903_LINEOUTVU;
- wm8903_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
- wm8903_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
- val = wm8903_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
+ val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
val |= WM8903_SPKVU;
- wm8903_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
- wm8903_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
+ snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
/* Enable DAC soft mute by default */
- val = wm8903_read(codec, WM8903_DAC_DIGITAL_1);
+ val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
val |= WM8903_DAC_MUTEMODE;
- wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
+ snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val);
wm8903_dai.dev = &i2c->dev;
wm8903_codec = codec;
@@ -1721,6 +1655,21 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8903_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8903_i2c_suspend NULL
+#define wm8903_i2c_resume NULL
+#endif
+
/* i2c codec control layer */
static const struct i2c_device_id wm8903_i2c_id[] = {
{ "wm8903", 0 },
@@ -1735,6 +1684,8 @@ static struct i2c_driver wm8903_i2c_driver = {
},
.probe = wm8903_i2c_probe,
.remove = __devexit_p(wm8903_i2c_remove),
+ .suspend = wm8903_i2c_suspend,
+ .resume = wm8903_i2c_resume,
.id_table = wm8903_i2c_id,
};
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b8e17d6..da97aae 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -106,50 +106,6 @@ static u16 wm8940_reg_defaults[] = {
0x0000, /* Mono Mixer Control */
};
-static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
-
- if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
- return -1;
-
- return cache[reg];
-}
-
-static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
-
- if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
- return -1;
-
- cache[reg] = value;
-
- return 0;
-}
-
-static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- int ret;
- u8 data[3] = { reg,
- (value & 0xff00) >> 8,
- (value & 0x00ff)
- };
-
- wm8940_write_reg_cache(codec, reg, value);
-
- ret = codec->hw_write(codec->control_data, data, 3);
-
- if (ret < 0)
- return ret;
- else if (ret != 3)
- return -EIO;
- return 0;
-}
-
static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
static const struct soc_enum wm8940_adc_companding_enum
= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
@@ -348,14 +304,14 @@ error_ret:
return ret;
}
-#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+#define wm8940_reset(c) snd_soc_write(c, WM8940_SOFTRESET, 0);
static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
- u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+ u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFE67;
+ u16 clk = snd_soc_read(codec, WM8940_CLOCK) & 0x1fe;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
@@ -366,7 +322,7 @@ static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
default:
return -EINVAL;
}
- wm8940_write(codec, WM8940_CLOCK, clk);
+ snd_soc_write(codec, WM8940_CLOCK, clk);
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
@@ -399,7 +355,7 @@ static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
break;
}
- wm8940_write(codec, WM8940_IFACE, iface);
+ snd_soc_write(codec, WM8940_IFACE, iface);
return 0;
}
@@ -411,9 +367,9 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
- u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
- u16 companding = wm8940_read_reg_cache(codec,
+ u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F;
+ u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1;
+ u16 companding = snd_soc_read(codec,
WM8940_COMPANDINGCTL) & 0xFFDF;
int ret;
@@ -442,7 +398,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_RATE_48000:
break;
}
- ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+ ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl);
if (ret)
goto error_ret;
@@ -462,10 +418,10 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
iface |= (3 << 5);
break;
}
- ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+ ret = snd_soc_write(codec, WM8940_COMPANDINGCTL, companding);
if (ret)
goto error_ret;
- ret = wm8940_write(codec, WM8940_IFACE, iface);
+ ret = snd_soc_write(codec, WM8940_IFACE, iface);
error_ret:
return ret;
@@ -474,19 +430,19 @@ error_ret:
static int wm8940_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+ u16 mute_reg = snd_soc_read(codec, WM8940_DAC) & 0xffbf;
if (mute)
mute_reg |= 0x40;
- return wm8940_write(codec, WM8940_DAC, mute_reg);
+ return snd_soc_write(codec, WM8940_DAC, mute_reg);
}
static int wm8940_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 val;
- u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+ u16 pwr_reg = snd_soc_read(codec, WM8940_POWER1) & 0x1F0;
int ret = 0;
switch (level) {
@@ -494,26 +450,26 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
/* ensure bufioen and biasen */
pwr_reg |= (1 << 2) | (1 << 3);
/* Enable thermal shutdown */
- val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
- ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+ val = snd_soc_read(codec, WM8940_OUTPUTCTL);
+ ret = snd_soc_write(codec, WM8940_OUTPUTCTL, val | 0x2);
if (ret)
break;
/* set vmid to 75k */
- ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1);
break;
case SND_SOC_BIAS_PREPARE:
/* ensure bufioen and biasen */
pwr_reg |= (1 << 2) | (1 << 3);
- ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1);
break;
case SND_SOC_BIAS_STANDBY:
/* ensure bufioen and biasen */
pwr_reg |= (1 << 2) | (1 << 3);
/* set vmid to 300k for standby */
- ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+ ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x2);
break;
case SND_SOC_BIAS_OFF:
- ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+ ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg);
break;
}
@@ -587,36 +543,36 @@ static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
u16 reg;
/* Turn off PLL */
- reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
- wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+ reg = snd_soc_read(codec, WM8940_POWER1);
+ snd_soc_write(codec, WM8940_POWER1, reg & 0x1df);
if (freq_in == 0 || freq_out == 0) {
/* Clock CODEC directly from MCLK */
- reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
- wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+ reg = snd_soc_read(codec, WM8940_CLOCK);
+ snd_soc_write(codec, WM8940_CLOCK, reg & 0x0ff);
/* Pll power down */
- wm8940_write(codec, WM8940_PLLN, (1 << 7));
+ snd_soc_write(codec, WM8940_PLLN, (1 << 7));
return 0;
}
/* Pll is followed by a frequency divide by 4 */
pll_factors(freq_out*4, freq_in);
if (pll_div.k)
- wm8940_write(codec, WM8940_PLLN,
+ snd_soc_write(codec, WM8940_PLLN,
(pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
else /* No factional component */
- wm8940_write(codec, WM8940_PLLN,
+ snd_soc_write(codec, WM8940_PLLN,
(pll_div.pre_scale << 4) | pll_div.n);
- wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
- wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
- wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+ snd_soc_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
/* Enable the PLL */
- reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
- wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+ reg = snd_soc_read(codec, WM8940_POWER1);
+ snd_soc_write(codec, WM8940_POWER1, reg | 0x020);
/* Run CODEC from PLL instead of MCLK */
- reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
- wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+ reg = snd_soc_read(codec, WM8940_CLOCK);
+ snd_soc_write(codec, WM8940_CLOCK, reg | 0x100);
return 0;
}
@@ -648,16 +604,16 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8940_BCLKDIV:
- reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
- ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+ reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3;
+ ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2));
break;
case WM8940_MCLKDIV:
- reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
- ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+ reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFF1F;
+ ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 5));
break;
case WM8940_OPCLKDIV:
- reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
- ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+ reg = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFCF;
+ ret = snd_soc_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
break;
}
return ret;
@@ -808,7 +764,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8940 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
-static int wm8940_register(struct wm8940_priv *wm8940)
+static int wm8940_register(struct wm8940_priv *wm8940,
+ enum snd_soc_control_type control)
{
struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
struct snd_soc_codec *codec = &wm8940->codec;
@@ -825,8 +782,6 @@ static int wm8940_register(struct wm8940_priv *wm8940)
codec->private_data = wm8940;
codec->name = "WM8940";
codec->owner = THIS_MODULE;
- codec->read = wm8940_read_reg_cache;
- codec->write = wm8940_write;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8940_set_bias_level;
codec->dai = &wm8940_dai;
@@ -834,6 +789,12 @@ static int wm8940_register(struct wm8940_priv *wm8940)
codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
codec->reg_cache = &wm8940->reg_cache;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
+ if (ret == 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
memcpy(codec->reg_cache, wm8940_reg_defaults,
sizeof(wm8940_reg_defaults));
@@ -847,15 +808,15 @@ static int wm8940_register(struct wm8940_priv *wm8940)
wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+ ret = snd_soc_write(codec, WM8940_POWER1, 0x180);
if (ret < 0)
return ret;
if (!pdata)
dev_warn(codec->dev, "No platform data supplied\n");
else {
- reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
- ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+ reg = snd_soc_read(codec, WM8940_OUTPUTCTL);
+ ret = snd_soc_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
if (ret < 0)
return ret;
}
@@ -904,7 +865,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c,
codec->control_data = i2c;
codec->dev = &i2c->dev;
- return wm8940_register(wm8940);
+ return wm8940_register(wm8940, SND_SOC_I2C);
}
static int __devexit wm8940_i2c_remove(struct i2c_client *client)
@@ -916,6 +877,21 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8940_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8940_i2c_suspend NULL
+#define wm8940_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8940_i2c_id[] = {
{ "wm8940", 0 },
{ }
@@ -929,6 +905,8 @@ static struct i2c_driver wm8940_i2c_driver = {
},
.probe = wm8940_i2c_probe,
.remove = __devexit_p(wm8940_i2c_remove),
+ .suspend = wm8940_i2c_suspend,
+ .resume = wm8940_i2c_resume,
.id_table = wm8940_i2c_id,
};
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index e224d8a..f59703b 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -69,61 +69,7 @@ struct wm8960_priv {
struct snd_soc_codec codec;
};
-/*
- * read wm8960 register cache
- */
-static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg == WM8960_RESET)
- return 0;
- if (reg >= WM8960_CACHEREGNUM)
- return -1;
- return cache[reg];
-}
-
-/*
- * write wm8960 register cache
- */
-static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
- u16 reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg >= WM8960_CACHEREGNUM)
- return;
- cache[reg] = value;
-}
-
-static inline unsigned int wm8960_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return wm8960_read_reg_cache(codec, reg);
-}
-
-/*
- * write to the WM8960 register space
- */
-static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8960 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8960_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0)
+#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0)
/* enumerated controls */
static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
@@ -420,7 +366,7 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
}
/* set iface */
- wm8960_write(codec, WM8960_IFACE1, iface);
+ snd_soc_write(codec, WM8960_IFACE1, iface);
return 0;
}
@@ -431,7 +377,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3;
+ u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
/* bit size */
switch (params_format(params)) {
@@ -446,19 +392,19 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
}
/* set iface */
- wm8960_write(codec, WM8960_IFACE1, iface);
+ snd_soc_write(codec, WM8960_IFACE1, iface);
return 0;
}
static int wm8960_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7;
+ u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7;
if (mute)
- wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
+ snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
else
- wm8960_write(codec, WM8960_DACCTL1, mute_reg);
+ snd_soc_write(codec, WM8960_DACCTL1, mute_reg);
return 0;
}
@@ -474,16 +420,16 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
/* Set VMID to 2x50k */
- reg = wm8960_read(codec, WM8960_POWER1);
+ reg = snd_soc_read(codec, WM8960_POWER1);
reg &= ~0x180;
reg |= 0x80;
- wm8960_write(codec, WM8960_POWER1, reg);
+ snd_soc_write(codec, WM8960_POWER1, reg);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Enable anti-pop features */
- wm8960_write(codec, WM8960_APOP1,
+ snd_soc_write(codec, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN | WM8960_BUFIOEN);
@@ -491,43 +437,43 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec,
reg = WM8960_DISOP;
if (pdata)
reg |= pdata->dres << 4;
- wm8960_write(codec, WM8960_APOP2, reg);
+ snd_soc_write(codec, WM8960_APOP2, reg);
msleep(400);
- wm8960_write(codec, WM8960_APOP2, 0);
+ snd_soc_write(codec, WM8960_APOP2, 0);
/* Enable & ramp VMID at 2x50k */
- reg = wm8960_read(codec, WM8960_POWER1);
+ reg = snd_soc_read(codec, WM8960_POWER1);
reg |= 0x80;
- wm8960_write(codec, WM8960_POWER1, reg);
+ snd_soc_write(codec, WM8960_POWER1, reg);
msleep(100);
/* Enable VREF */
- wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF);
+ snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF);
/* Disable anti-pop features */
- wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
+ snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
}
/* Set VMID to 2x250k */
- reg = wm8960_read(codec, WM8960_POWER1);
+ reg = snd_soc_read(codec, WM8960_POWER1);
reg &= ~0x180;
reg |= 0x100;
- wm8960_write(codec, WM8960_POWER1, reg);
+ snd_soc_write(codec, WM8960_POWER1, reg);
break;
case SND_SOC_BIAS_OFF:
/* Enable anti-pop features */
- wm8960_write(codec, WM8960_APOP1,
+ snd_soc_write(codec, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN | WM8960_BUFIOEN);
/* Disable VMID and VREF, let them discharge */
- wm8960_write(codec, WM8960_POWER1, 0);
+ snd_soc_write(codec, WM8960_POWER1, 0);
msleep(600);
- wm8960_write(codec, WM8960_APOP1, 0);
+ snd_soc_write(codec, WM8960_APOP1, 0);
break;
}
@@ -610,33 +556,33 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
/* Disable the PLL: even if we are changing the frequency the
* PLL needs to be disabled while we do so. */
- wm8960_write(codec, WM8960_CLOCK1,
- wm8960_read(codec, WM8960_CLOCK1) & ~1);
- wm8960_write(codec, WM8960_POWER2,
- wm8960_read(codec, WM8960_POWER2) & ~1);
+ snd_soc_write(codec, WM8960_CLOCK1,
+ snd_soc_read(codec, WM8960_CLOCK1) & ~1);
+ snd_soc_write(codec, WM8960_POWER2,
+ snd_soc_read(codec, WM8960_POWER2) & ~1);
if (!freq_in || !freq_out)
return 0;
- reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f;
+ reg = snd_soc_read(codec, WM8960_PLL1) & ~0x3f;
reg |= pll_div.pre_div << 4;
reg |= pll_div.n;
if (pll_div.k) {
reg |= 0x20;
- wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
- wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
- wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
+ snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
}
- wm8960_write(codec, WM8960_PLL1, reg);
+ snd_soc_write(codec, WM8960_PLL1, reg);
/* Turn it on */
- wm8960_write(codec, WM8960_POWER2,
- wm8960_read(codec, WM8960_POWER2) | 1);
+ snd_soc_write(codec, WM8960_POWER2,
+ snd_soc_read(codec, WM8960_POWER2) | 1);
msleep(250);
- wm8960_write(codec, WM8960_CLOCK1,
- wm8960_read(codec, WM8960_CLOCK1) | 1);
+ snd_soc_write(codec, WM8960_CLOCK1,
+ snd_soc_read(codec, WM8960_CLOCK1) | 1);
return 0;
}
@@ -649,28 +595,28 @@ static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8960_SYSCLKSEL:
- reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe;
- wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1fe;
+ snd_soc_write(codec, WM8960_CLOCK1, reg | div);
break;
case WM8960_SYSCLKDIV:
- reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9;
- wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1f9;
+ snd_soc_write(codec, WM8960_CLOCK1, reg | div);
break;
case WM8960_DACDIV:
- reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7;
- wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1c7;
+ snd_soc_write(codec, WM8960_CLOCK1, reg | div);
break;
case WM8960_OPCLKDIV:
- reg = wm8960_read(codec, WM8960_PLL1) & 0x03f;
- wm8960_write(codec, WM8960_PLL1, reg | div);
+ reg = snd_soc_read(codec, WM8960_PLL1) & 0x03f;
+ snd_soc_write(codec, WM8960_PLL1, reg | div);
break;
case WM8960_DCLKDIV:
- reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f;
- wm8960_write(codec, WM8960_CLOCK2, reg | div);
+ reg = snd_soc_read(codec, WM8960_CLOCK2) & 0x03f;
+ snd_soc_write(codec, WM8960_CLOCK2, reg | div);
break;
case WM8960_TOCLKSEL:
- reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd;
- wm8960_write(codec, WM8960_ADDCTL1, reg | div);
+ reg = snd_soc_read(codec, WM8960_ADDCTL1) & 0x1fd;
+ snd_soc_write(codec, WM8960_ADDCTL1, reg | div);
break;
default:
return -EINVAL;
@@ -801,7 +747,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8960 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960);
-static int wm8960_register(struct wm8960_priv *wm8960)
+static int wm8960_register(struct wm8960_priv *wm8960,
+ enum snd_soc_control_type control)
{
struct wm8960_data *pdata = wm8960->codec.dev->platform_data;
struct snd_soc_codec *codec = &wm8960->codec;
@@ -810,7 +757,8 @@ static int wm8960_register(struct wm8960_priv *wm8960)
if (wm8960_codec) {
dev_err(codec->dev, "Another WM8960 is registered\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto err;
}
if (!pdata) {
@@ -829,8 +777,6 @@ static int wm8960_register(struct wm8960_priv *wm8960)
codec->private_data = wm8960;
codec->name = "WM8960";
codec->owner = THIS_MODULE;
- codec->read = wm8960_read_reg_cache;
- codec->write = wm8960_write;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm8960_set_bias_level;
codec->dai = &wm8960_dai;
@@ -840,10 +786,16 @@ static int wm8960_register(struct wm8960_priv *wm8960)
memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg));
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
ret = wm8960_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
- return ret;
+ goto err;
}
wm8960_dai.dev = codec->dev;
@@ -851,43 +803,48 @@ static int wm8960_register(struct wm8960_priv *wm8960)
wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Latch the update bits */
- reg = wm8960_read(codec, WM8960_LINVOL);
- wm8960_write(codec, WM8960_LINVOL, reg | 0x100);
- reg = wm8960_read(codec, WM8960_RINVOL);
- wm8960_write(codec, WM8960_RINVOL, reg | 0x100);
- reg = wm8960_read(codec, WM8960_LADC);
- wm8960_write(codec, WM8960_LADC, reg | 0x100);
- reg = wm8960_read(codec, WM8960_RADC);
- wm8960_write(codec, WM8960_RADC, reg | 0x100);
- reg = wm8960_read(codec, WM8960_LDAC);
- wm8960_write(codec, WM8960_LDAC, reg | 0x100);
- reg = wm8960_read(codec, WM8960_RDAC);
- wm8960_write(codec, WM8960_RDAC, reg | 0x100);
- reg = wm8960_read(codec, WM8960_LOUT1);
- wm8960_write(codec, WM8960_LOUT1, reg | 0x100);
- reg = wm8960_read(codec, WM8960_ROUT1);
- wm8960_write(codec, WM8960_ROUT1, reg | 0x100);
- reg = wm8960_read(codec, WM8960_LOUT2);
- wm8960_write(codec, WM8960_LOUT2, reg | 0x100);
- reg = wm8960_read(codec, WM8960_ROUT2);
- wm8960_write(codec, WM8960_ROUT2, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_LINVOL);
+ snd_soc_write(codec, WM8960_LINVOL, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_RINVOL);
+ snd_soc_write(codec, WM8960_RINVOL, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_LADC);
+ snd_soc_write(codec, WM8960_LADC, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_RADC);
+ snd_soc_write(codec, WM8960_RADC, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_LDAC);
+ snd_soc_write(codec, WM8960_LDAC, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_RDAC);
+ snd_soc_write(codec, WM8960_RDAC, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_LOUT1);
+ snd_soc_write(codec, WM8960_LOUT1, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_ROUT1);
+ snd_soc_write(codec, WM8960_ROUT1, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_LOUT2);
+ snd_soc_write(codec, WM8960_LOUT2, reg | 0x100);
+ reg = snd_soc_read(codec, WM8960_ROUT2);
+ snd_soc_write(codec, WM8960_ROUT2, reg | 0x100);
wm8960_codec = codec;
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
- return ret;
+ goto err;
}
ret = snd_soc_register_dai(&wm8960_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
- snd_soc_unregister_codec(codec);
- return ret;
+ goto err_codec;
}
return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8960);
+ return ret;
}
static void wm8960_unregister(struct wm8960_priv *wm8960)
@@ -910,14 +867,13 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
codec = &wm8960->codec;
- codec->hw_write = (hw_write_t)i2c_master_send;
i2c_set_clientdata(i2c, wm8960);
codec->control_data = i2c;
codec->dev = &i2c->dev;
- return wm8960_register(wm8960);
+ return wm8960_register(wm8960, SND_SOC_I2C);
}
static __devexit int wm8960_i2c_remove(struct i2c_client *client)
@@ -927,6 +883,21 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8960_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8960_i2c_suspend NULL
+#define wm8960_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8960_i2c_id[] = {
{ "wm8960", 0 },
{ }
@@ -940,6 +911,8 @@ static struct i2c_driver wm8960_i2c_driver = {
},
.probe = wm8960_i2c_probe,
.remove = __devexit_p(wm8960_i2c_remove),
+ .suspend = wm8960_i2c_suspend,
+ .resume = wm8960_i2c_resume,
.id_table = wm8960_i2c_id,
};
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
new file mode 100644
index 0000000..5030320
--- /dev/null
+++ b/sound/soc/codecs/wm8961.c
@@ -0,0 +1,1265 @@
+/*
+ * wm8961.c -- WM8961 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Currently unimplemented features:
+ * - ALC
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8961.h"
+
+#define WM8961_MAX_REGISTER 0xFC
+
+static u16 wm8961_reg_defaults[] = {
+ 0x009F, /* R0 - Left Input volume */
+ 0x009F, /* R1 - Right Input volume */
+ 0x0000, /* R2 - LOUT1 volume */
+ 0x0000, /* R3 - ROUT1 volume */
+ 0x0020, /* R4 - Clocking1 */
+ 0x0008, /* R5 - ADC & DAC Control 1 */
+ 0x0000, /* R6 - ADC & DAC Control 2 */
+ 0x000A, /* R7 - Audio Interface 0 */
+ 0x01F4, /* R8 - Clocking2 */
+ 0x0000, /* R9 - Audio Interface 1 */
+ 0x00FF, /* R10 - Left DAC volume */
+ 0x00FF, /* R11 - Right DAC volume */
+ 0x0000, /* R12 */
+ 0x0000, /* R13 */
+ 0x0040, /* R14 - Audio Interface 2 */
+ 0x0000, /* R15 - Software Reset */
+ 0x0000, /* R16 */
+ 0x007B, /* R17 - ALC1 */
+ 0x0000, /* R18 - ALC2 */
+ 0x0032, /* R19 - ALC3 */
+ 0x0000, /* R20 - Noise Gate */
+ 0x00C0, /* R21 - Left ADC volume */
+ 0x00C0, /* R22 - Right ADC volume */
+ 0x0120, /* R23 - Additional control(1) */
+ 0x0000, /* R24 - Additional control(2) */
+ 0x0000, /* R25 - Pwr Mgmt (1) */
+ 0x0000, /* R26 - Pwr Mgmt (2) */
+ 0x0000, /* R27 - Additional Control (3) */
+ 0x0000, /* R28 - Anti-pop */
+ 0x0000, /* R29 */
+ 0x005F, /* R30 - Clocking 3 */
+ 0x0000, /* R31 */
+ 0x0000, /* R32 - ADCL signal path */
+ 0x0000, /* R33 - ADCR signal path */
+ 0x0000, /* R34 */
+ 0x0000, /* R35 */
+ 0x0000, /* R36 */
+ 0x0000, /* R37 */
+ 0x0000, /* R38 */
+ 0x0000, /* R39 */
+ 0x0000, /* R40 - LOUT2 volume */
+ 0x0000, /* R41 - ROUT2 volume */
+ 0x0000, /* R42 */
+ 0x0000, /* R43 */
+ 0x0000, /* R44 */
+ 0x0000, /* R45 */
+ 0x0000, /* R46 */
+ 0x0000, /* R47 - Pwr Mgmt (3) */
+ 0x0023, /* R48 - Additional Control (4) */
+ 0x0000, /* R49 - Class D Control 1 */
+ 0x0000, /* R50 */
+ 0x0003, /* R51 - Class D Control 2 */
+ 0x0000, /* R52 */
+ 0x0000, /* R53 */
+ 0x0000, /* R54 */
+ 0x0000, /* R55 */
+ 0x0106, /* R56 - Clocking 4 */
+ 0x0000, /* R57 - DSP Sidetone 0 */
+ 0x0000, /* R58 - DSP Sidetone 1 */
+ 0x0000, /* R59 */
+ 0x0000, /* R60 - DC Servo 0 */
+ 0x0000, /* R61 - DC Servo 1 */
+ 0x0000, /* R62 */
+ 0x015E, /* R63 - DC Servo 3 */
+ 0x0010, /* R64 */
+ 0x0010, /* R65 - DC Servo 5 */
+ 0x0000, /* R66 */
+ 0x0001, /* R67 */
+ 0x0003, /* R68 - Analogue PGA Bias */
+ 0x0000, /* R69 - Analogue HP 0 */
+ 0x0060, /* R70 */
+ 0x01FB, /* R71 - Analogue HP 2 */
+ 0x0000, /* R72 - Charge Pump 1 */
+ 0x0065, /* R73 */
+ 0x005F, /* R74 */
+ 0x0059, /* R75 */
+ 0x006B, /* R76 */
+ 0x0038, /* R77 */
+ 0x000C, /* R78 */
+ 0x000A, /* R79 */
+ 0x006B, /* R80 */
+ 0x0000, /* R81 */
+ 0x0000, /* R82 - Charge Pump B */
+ 0x0087, /* R83 */
+ 0x0000, /* R84 */
+ 0x005C, /* R85 */
+ 0x0000, /* R86 */
+ 0x0000, /* R87 - Write Sequencer 1 */
+ 0x0000, /* R88 - Write Sequencer 2 */
+ 0x0000, /* R89 - Write Sequencer 3 */
+ 0x0000, /* R90 - Write Sequencer 4 */
+ 0x0000, /* R91 - Write Sequencer 5 */
+ 0x0000, /* R92 - Write Sequencer 6 */
+ 0x0000, /* R93 - Write Sequencer 7 */
+ 0x0000, /* R94 */
+ 0x0000, /* R95 */
+ 0x0000, /* R96 */
+ 0x0000, /* R97 */
+ 0x0000, /* R98 */
+ 0x0000, /* R99 */
+ 0x0000, /* R100 */
+ 0x0000, /* R101 */
+ 0x0000, /* R102 */
+ 0x0000, /* R103 */
+ 0x0000, /* R104 */
+ 0x0000, /* R105 */
+ 0x0000, /* R106 */
+ 0x0000, /* R107 */
+ 0x0000, /* R108 */
+ 0x0000, /* R109 */
+ 0x0000, /* R110 */
+ 0x0000, /* R111 */
+ 0x0000, /* R112 */
+ 0x0000, /* R113 */
+ 0x0000, /* R114 */
+ 0x0000, /* R115 */
+ 0x0000, /* R116 */
+ 0x0000, /* R117 */
+ 0x0000, /* R118 */
+ 0x0000, /* R119 */
+ 0x0000, /* R120 */
+ 0x0000, /* R121 */
+ 0x0000, /* R122 */
+ 0x0000, /* R123 */
+ 0x0000, /* R124 */
+ 0x0000, /* R125 */
+ 0x0000, /* R126 */
+ 0x0000, /* R127 */
+ 0x0000, /* R128 */
+ 0x0000, /* R129 */
+ 0x0000, /* R130 */
+ 0x0000, /* R131 */
+ 0x0000, /* R132 */
+ 0x0000, /* R133 */
+ 0x0000, /* R134 */
+ 0x0000, /* R135 */
+ 0x0000, /* R136 */
+ 0x0000, /* R137 */
+ 0x0000, /* R138 */
+ 0x0000, /* R139 */
+ 0x0000, /* R140 */
+ 0x0000, /* R141 */
+ 0x0000, /* R142 */
+ 0x0000, /* R143 */
+ 0x0000, /* R144 */
+ 0x0000, /* R145 */
+ 0x0000, /* R146 */
+ 0x0000, /* R147 */
+ 0x0000, /* R148 */
+ 0x0000, /* R149 */
+ 0x0000, /* R150 */
+ 0x0000, /* R151 */
+ 0x0000, /* R152 */
+ 0x0000, /* R153 */
+ 0x0000, /* R154 */
+ 0x0000, /* R155 */
+ 0x0000, /* R156 */
+ 0x0000, /* R157 */
+ 0x0000, /* R158 */
+ 0x0000, /* R159 */
+ 0x0000, /* R160 */
+ 0x0000, /* R161 */
+ 0x0000, /* R162 */
+ 0x0000, /* R163 */
+ 0x0000, /* R164 */
+ 0x0000, /* R165 */
+ 0x0000, /* R166 */
+ 0x0000, /* R167 */
+ 0x0000, /* R168 */
+ 0x0000, /* R169 */
+ 0x0000, /* R170 */
+ 0x0000, /* R171 */
+ 0x0000, /* R172 */
+ 0x0000, /* R173 */
+ 0x0000, /* R174 */
+ 0x0000, /* R175 */
+ 0x0000, /* R176 */
+ 0x0000, /* R177 */
+ 0x0000, /* R178 */
+ 0x0000, /* R179 */
+ 0x0000, /* R180 */
+ 0x0000, /* R181 */
+ 0x0000, /* R182 */
+ 0x0000, /* R183 */
+ 0x0000, /* R184 */
+ 0x0000, /* R185 */
+ 0x0000, /* R186 */
+ 0x0000, /* R187 */
+ 0x0000, /* R188 */
+ 0x0000, /* R189 */
+ 0x0000, /* R190 */
+ 0x0000, /* R191 */
+ 0x0000, /* R192 */
+ 0x0000, /* R193 */
+ 0x0000, /* R194 */
+ 0x0000, /* R195 */
+ 0x0030, /* R196 */
+ 0x0006, /* R197 */
+ 0x0000, /* R198 */
+ 0x0060, /* R199 */
+ 0x0000, /* R200 */
+ 0x003F, /* R201 */
+ 0x0000, /* R202 */
+ 0x0000, /* R203 */
+ 0x0000, /* R204 */
+ 0x0001, /* R205 */
+ 0x0000, /* R206 */
+ 0x0181, /* R207 */
+ 0x0005, /* R208 */
+ 0x0008, /* R209 */
+ 0x0008, /* R210 */
+ 0x0000, /* R211 */
+ 0x013B, /* R212 */
+ 0x0000, /* R213 */
+ 0x0000, /* R214 */
+ 0x0000, /* R215 */
+ 0x0000, /* R216 */
+ 0x0070, /* R217 */
+ 0x0000, /* R218 */
+ 0x0000, /* R219 */
+ 0x0000, /* R220 */
+ 0x0000, /* R221 */
+ 0x0000, /* R222 */
+ 0x0003, /* R223 */
+ 0x0000, /* R224 */
+ 0x0000, /* R225 */
+ 0x0001, /* R226 */
+ 0x0008, /* R227 */
+ 0x0000, /* R228 */
+ 0x0000, /* R229 */
+ 0x0000, /* R230 */
+ 0x0000, /* R231 */
+ 0x0004, /* R232 */
+ 0x0000, /* R233 */
+ 0x0000, /* R234 */
+ 0x0000, /* R235 */
+ 0x0000, /* R236 */
+ 0x0000, /* R237 */
+ 0x0080, /* R238 */
+ 0x0000, /* R239 */
+ 0x0000, /* R240 */
+ 0x0000, /* R241 */
+ 0x0000, /* R242 */
+ 0x0000, /* R243 */
+ 0x0000, /* R244 */
+ 0x0052, /* R245 */
+ 0x0110, /* R246 */
+ 0x0040, /* R247 */
+ 0x0000, /* R248 */
+ 0x0030, /* R249 */
+ 0x0000, /* R250 */
+ 0x0000, /* R251 */
+ 0x0001, /* R252 - General test 1 */
+};
+
+struct wm8961_priv {
+ struct snd_soc_codec codec;
+ int sysclk;
+ u16 reg_cache[WM8961_MAX_REGISTER];
+};
+
+static int wm8961_volatile_register(unsigned int reg)
+{
+ switch (reg) {
+ case WM8961_SOFTWARE_RESET:
+ case WM8961_WRITE_SEQUENCER_7:
+ case WM8961_DC_SERVO_1:
+ return 1;
+
+ default:
+ return 0;
+ }
+}
+
+static int wm8961_reset(struct snd_soc_codec *codec)
+{
+ return snd_soc_write(codec, WM8961_SOFTWARE_RESET, 0);
+}
+
+/*
+ * The headphone output supports special anti-pop sequences giving
+ * silent power up and power down.
+ */
+static int wm8961_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 hp_reg = snd_soc_read(codec, WM8961_ANALOGUE_HP_0);
+ u16 cp_reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_1);
+ u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
+ u16 dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1);
+ int timeout = 500;
+
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ /* Make sure the output is shorted */
+ hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Enable the charge pump */
+ cp_reg |= WM8961_CP_ENA;
+ snd_soc_write(codec, WM8961_CHARGE_PUMP_1, cp_reg);
+ mdelay(5);
+
+ /* Enable the PGA */
+ pwr_reg |= WM8961_LOUT1_PGA | WM8961_ROUT1_PGA;
+ snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
+
+ /* Enable the amplifier */
+ hp_reg |= WM8961_HPR_ENA | WM8961_HPL_ENA;
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Second stage enable */
+ hp_reg |= WM8961_HPR_ENA_DLY | WM8961_HPL_ENA_DLY;
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Enable the DC servo & trigger startup */
+ dcs_reg |=
+ WM8961_DCS_ENA_CHAN_HPR | WM8961_DCS_TRIG_STARTUP_HPR |
+ WM8961_DCS_ENA_CHAN_HPL | WM8961_DCS_TRIG_STARTUP_HPL;
+ dev_dbg(codec->dev, "Enabling DC servo\n");
+
+ snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg);
+ do {
+ msleep(1);
+ dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1);
+ } while (--timeout &&
+ dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
+ WM8961_DCS_TRIG_STARTUP_HPL));
+ if (dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
+ WM8961_DCS_TRIG_STARTUP_HPL))
+ dev_err(codec->dev, "DC servo timed out\n");
+ else
+ dev_dbg(codec->dev, "DC servo startup complete\n");
+
+ /* Enable the output stage */
+ hp_reg |= WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP;
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Remove the short on the output stage */
+ hp_reg |= WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT;
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+ }
+
+ if (event & SND_SOC_DAPM_PRE_PMD) {
+ /* Short the output */
+ hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Disable the output stage */
+ hp_reg &= ~(WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP);
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Disable DC offset cancellation */
+ dcs_reg &= ~(WM8961_DCS_ENA_CHAN_HPR |
+ WM8961_DCS_ENA_CHAN_HPL);
+ snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg);
+
+ /* Finish up */
+ hp_reg &= ~(WM8961_HPR_ENA_DLY | WM8961_HPR_ENA |
+ WM8961_HPL_ENA_DLY | WM8961_HPL_ENA);
+ snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
+
+ /* Disable the PGA */
+ pwr_reg &= ~(WM8961_LOUT1_PGA | WM8961_ROUT1_PGA);
+ snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
+
+ /* Disable the charge pump */
+ dev_dbg(codec->dev, "Disabling charge pump\n");
+ snd_soc_write(codec, WM8961_CHARGE_PUMP_1,
+ cp_reg & ~WM8961_CP_ENA);
+ }
+
+ return 0;
+}
+
+static int wm8961_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
+ u16 spk_reg = snd_soc_read(codec, WM8961_CLASS_D_CONTROL_1);
+
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ /* Enable the PGA */
+ pwr_reg |= WM8961_SPKL_PGA | WM8961_SPKR_PGA;
+ snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
+
+ /* Enable the amplifier */
+ spk_reg |= WM8961_SPKL_ENA | WM8961_SPKR_ENA;
+ snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg);
+ }
+
+ if (event & SND_SOC_DAPM_PRE_PMD) {
+ /* Enable the amplifier */
+ spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA);
+ snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg);
+
+ /* Enable the PGA */
+ pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA);
+ snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
+ }
+
+ return 0;
+}
+
+static const char *adc_hpf_text[] = {
+ "Hi-fi", "Voice 1", "Voice 2", "Voice 3",
+};
+
+static const struct soc_enum adc_hpf =
+ SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text);
+
+static const char *dac_deemph_text[] = {
+ "None", "32kHz", "44.1kHz", "48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+ SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
+static unsigned int boost_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(13, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(20, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(29, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -2325, 75, 0);
+
+static const struct snd_kcontrol_new wm8961_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Headphone Volume", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_TLV("Headphone Secondary Volume", WM8961_ANALOGUE_HP_2,
+ 6, 3, 7, 0, hp_sec_tlv),
+SOC_DOUBLE_R("Headphone ZC Switch", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Speaker Volume", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Speaker ZC Switch", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
+ 7, 1, 0),
+SOC_SINGLE("Speaker AC Gain", WM8961_CLASS_D_CONTROL_2, 0, 7, 0),
+
+SOC_SINGLE("DAC x128 OSR Switch", WM8961_ADC_DAC_CONTROL_2, 0, 1, 0),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+SOC_SINGLE("DAC Soft Mute Switch", WM8961_ADC_DAC_CONTROL_2, 3, 1, 0),
+
+SOC_DOUBLE_R_TLV("Sidetone Volume", WM8961_DSP_SIDETONE_0,
+ WM8961_DSP_SIDETONE_1, 4, 12, 0, sidetone_tlv),
+
+SOC_SINGLE("ADC High Pass Filter Switch", WM8961_ADC_DAC_CONTROL_1, 0, 1, 0),
+SOC_ENUM("ADC High Pass Filter Mode", adc_hpf),
+
+SOC_DOUBLE_R_TLV("Capture Volume",
+ WM8961_LEFT_ADC_VOLUME, WM8961_RIGHT_ADC_VOLUME,
+ 1, 119, 0, adc_tlv),
+SOC_DOUBLE_R_TLV("Capture Boost Volume",
+ WM8961_ADCL_SIGNAL_PATH, WM8961_ADCR_SIGNAL_PATH,
+ 4, 3, 0, boost_tlv),
+SOC_DOUBLE_R_TLV("Capture PGA Volume",
+ WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
+ 0, 62, 0, pga_tlv),
+SOC_DOUBLE_R("Capture PGA ZC Switch",
+ WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
+ 6, 1, 1),
+SOC_DOUBLE_R("Capture PGA Switch",
+ WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
+ 7, 1, 1),
+};
+
+static const char *sidetone_text[] = {
+ "None", "Left", "Right"
+};
+
+static const struct soc_enum dacl_sidetone =
+ SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text);
+
+static const struct soc_enum dacr_sidetone =
+ SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text);
+
+static const struct snd_kcontrol_new dacl_mux =
+ SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone);
+
+static const struct snd_kcontrol_new dacr_mux =
+ SOC_DAPM_ENUM("DACR Sidetone", dacr_sidetone);
+
+static const struct snd_soc_dapm_widget wm8961_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("LINPUT"),
+SND_SOC_DAPM_INPUT("RINPUT"),
+
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8961_CLOCKING2, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Left Input", WM8961_PWR_MGMT_1, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0),
+SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0),
+
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8961_PWR_MGMT_1, 1, 0),
+
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux),
+
+SND_SOC_DAPM_DAC("DACL", "HiFi Playback", WM8961_PWR_MGMT_2, 8, 0),
+SND_SOC_DAPM_DAC("DACR", "HiFi Playback", WM8961_PWR_MGMT_2, 7, 0),
+
+/* Handle as a mono path for DCS */
+SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM,
+ 4, 0, NULL, 0, wm8961_hp_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_E("Speaker Output", SND_SOC_NOPM,
+ 4, 0, NULL, 0, wm8961_spk_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+SND_SOC_DAPM_OUTPUT("SPK_LN"),
+SND_SOC_DAPM_OUTPUT("SPK_LP"),
+SND_SOC_DAPM_OUTPUT("SPK_RN"),
+SND_SOC_DAPM_OUTPUT("SPK_RP"),
+};
+
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "DACL", NULL, "CLK_DSP" },
+ { "DACL", NULL, "DACL Sidetone" },
+ { "DACR", NULL, "CLK_DSP" },
+ { "DACR", NULL, "DACR Sidetone" },
+
+ { "DACL Sidetone", "Left", "ADCL" },
+ { "DACL Sidetone", "Right", "ADCR" },
+
+ { "DACR Sidetone", "Left", "ADCL" },
+ { "DACR Sidetone", "Right", "ADCR" },
+
+ { "HP_L", NULL, "Headphone Output" },
+ { "HP_R", NULL, "Headphone Output" },
+ { "Headphone Output", NULL, "DACL" },
+ { "Headphone Output", NULL, "DACR" },
+
+ { "SPK_LN", NULL, "Speaker Output" },
+ { "SPK_LP", NULL, "Speaker Output" },
+ { "SPK_RN", NULL, "Speaker Output" },
+ { "SPK_RP", NULL, "Speaker Output" },
+
+ { "Speaker Output", NULL, "DACL" },
+ { "Speaker Output", NULL, "DACR" },
+
+ { "ADCL", NULL, "Left Input" },
+ { "ADCL", NULL, "CLK_DSP" },
+ { "ADCR", NULL, "Right Input" },
+ { "ADCR", NULL, "CLK_DSP" },
+
+ { "Left Input", NULL, "LINPUT" },
+ { "Right Input", NULL, "RINPUT" },
+
+};
+
+/* Values for CLK_SYS_RATE */
+static struct {
+ int ratio;
+ u16 val;
+} wm8961_clk_sys_ratio[] = {
+ { 64, 0 },
+ { 128, 1 },
+ { 192, 2 },
+ { 256, 3 },
+ { 384, 4 },
+ { 512, 5 },
+ { 768, 6 },
+ { 1024, 7 },
+ { 1408, 8 },
+ { 1536, 9 },
+};
+
+/* Values for SAMPLE_RATE */
+static struct {
+ int rate;
+ u16 val;
+} wm8961_srate[] = {
+ { 48000, 0 },
+ { 44100, 0 },
+ { 32000, 1 },
+ { 22050, 2 },
+ { 24000, 2 },
+ { 16000, 3 },
+ { 11250, 4 },
+ { 12000, 4 },
+ { 8000, 5 },
+};
+
+static int wm8961_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8961_priv *wm8961 = codec->private_data;
+ int i, best, target, fs;
+ u16 reg;
+
+ fs = params_rate(params);
+
+ if (!wm8961->sysclk) {
+ dev_err(codec->dev, "MCLK has not been specified\n");
+ return -EINVAL;
+ }
+
+ /* Find the closest sample rate for the filters */
+ best = 0;
+ for (i = 0; i < ARRAY_SIZE(wm8961_srate); i++) {
+ if (abs(wm8961_srate[i].rate - fs) <
+ abs(wm8961_srate[best].rate - fs))
+ best = i;
+ }
+ reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_3);
+ reg &= ~WM8961_SAMPLE_RATE_MASK;
+ reg |= wm8961_srate[best].val;
+ snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_3, reg);
+ dev_dbg(codec->dev, "Selected SRATE %dHz for %dHz\n",
+ wm8961_srate[best].rate, fs);
+
+ /* Select a CLK_SYS/fs ratio equal to or higher than required */
+ target = wm8961->sysclk / fs;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) {
+ dev_err(codec->dev,
+ "SYSCLK must be at least 64*fs for DAC\n");
+ return -EINVAL;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) {
+ dev_err(codec->dev,
+ "SYSCLK must be at least 256*fs for ADC\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm8961_clk_sys_ratio); i++) {
+ if (wm8961_clk_sys_ratio[i].ratio >= target)
+ break;
+ }
+ if (i == ARRAY_SIZE(wm8961_clk_sys_ratio)) {
+ dev_err(codec->dev, "Unable to generate CLK_SYS_RATE\n");
+ return -EINVAL;
+ }
+ dev_dbg(codec->dev, "Selected CLK_SYS_RATE of %d for %d/%d=%d\n",
+ wm8961_clk_sys_ratio[i].ratio, wm8961->sysclk, fs,
+ wm8961->sysclk / fs);
+
+ reg = snd_soc_read(codec, WM8961_CLOCKING_4);
+ reg &= ~WM8961_CLK_SYS_RATE_MASK;
+ reg |= wm8961_clk_sys_ratio[i].val << WM8961_CLK_SYS_RATE_SHIFT;
+ snd_soc_write(codec, WM8961_CLOCKING_4, reg);
+
+ reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0);
+ reg &= ~WM8961_WL_MASK;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ reg |= 1 << WM8961_WL_SHIFT;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ reg |= 2 << WM8961_WL_SHIFT;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ reg |= 3 << WM8961_WL_SHIFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, reg);
+
+ /* Sloping stop-band filter is recommended for <= 24kHz */
+ reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2);
+ if (fs <= 24000)
+ reg |= WM8961_DACSLOPE;
+ else
+ reg &= WM8961_DACSLOPE;
+ snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
+
+ return 0;
+}
+
+static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq,
+ int dir)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8961_priv *wm8961 = codec->private_data;
+ u16 reg = snd_soc_read(codec, WM8961_CLOCKING1);
+
+ if (freq > 33000000) {
+ dev_err(codec->dev, "MCLK must be <33MHz\n");
+ return -EINVAL;
+ }
+
+ if (freq > 16500000) {
+ dev_dbg(codec->dev, "Using MCLK/2 for %dHz MCLK\n", freq);
+ reg |= WM8961_MCLKDIV;
+ freq /= 2;
+ } else {
+ dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
+ reg &= WM8961_MCLKDIV;
+ }
+
+ snd_soc_write(codec, WM8961_CLOCKING1, reg);
+
+ wm8961->sysclk = freq;
+
+ return 0;
+}
+
+static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 aif = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0);
+
+ aif &= ~(WM8961_BCLKINV | WM8961_LRP |
+ WM8961_MS | WM8961_FORMAT_MASK);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif |= WM8961_MS;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif |= 1;
+ break;
+
+ case SND_SOC_DAIFMT_I2S:
+ aif |= 2;
+ break;
+
+ case SND_SOC_DAIFMT_DSP_B:
+ aif |= WM8961_LRP;
+ case SND_SOC_DAIFMT_DSP_A:
+ aif |= 3;
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif |= WM8961_LRP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif |= WM8961_BCLKINV;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif |= WM8961_BCLKINV | WM8961_LRP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, aif);
+}
+
+static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_2);
+
+ if (tristate)
+ reg |= WM8961_TRIS;
+ else
+ reg &= ~WM8961_TRIS;
+
+ return snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_2, reg);
+}
+
+static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_1);
+
+ if (mute)
+ reg |= WM8961_DACMU;
+ else
+ reg &= ~WM8961_DACMU;
+
+ msleep(17);
+
+ return snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_1, reg);
+}
+
+static int wm8961_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8961_BCLK:
+ reg = snd_soc_read(codec, WM8961_CLOCKING2);
+ reg &= ~WM8961_BCLKDIV_MASK;
+ reg |= div;
+ snd_soc_write(codec, WM8961_CLOCKING2, reg);
+ break;
+
+ case WM8961_LRCLK:
+ reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_2);
+ reg &= ~WM8961_LRCLK_RATE_MASK;
+ reg |= div;
+ snd_soc_write(codec, WM8961_AUDIO_INTERFACE_2, reg);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8961_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+
+ /* This is all slightly unusual since we have no bypass paths
+ * and the output amplifier structure means we can just slam
+ * the biases straight up rather than having to ramp them
+ * slowly.
+ */
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+ /* Enable bias generation */
+ reg = snd_soc_read(codec, WM8961_ANTI_POP);
+ reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
+ snd_soc_write(codec, WM8961_ANTI_POP, reg);
+
+ /* VMID=2*50k, VREF */
+ reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
+ reg &= ~WM8961_VMIDSEL_MASK;
+ reg |= (1 << WM8961_VMIDSEL_SHIFT) | WM8961_VREF;
+ snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
+ }
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_PREPARE) {
+ /* VREF off */
+ reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
+ reg &= ~WM8961_VREF;
+ snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
+
+ /* Bias generation off */
+ reg = snd_soc_read(codec, WM8961_ANTI_POP);
+ reg &= ~(WM8961_BUFIOEN | WM8961_BUFDCOPEN);
+ snd_soc_write(codec, WM8961_ANTI_POP, reg);
+
+ /* VMID off */
+ reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
+ reg &= ~WM8961_VMIDSEL_MASK;
+ snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+
+#define WM8961_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8961_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8961_dai_ops = {
+ .hw_params = wm8961_hw_params,
+ .set_sysclk = wm8961_set_sysclk,
+ .set_fmt = wm8961_set_fmt,
+ .digital_mute = wm8961_digital_mute,
+ .set_tristate = wm8961_set_tristate,
+ .set_clkdiv = wm8961_set_clkdiv,
+};
+
+struct snd_soc_dai wm8961_dai = {
+ .name = "WM8961",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8961_RATES,
+ .formats = WM8961_FORMATS,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8961_RATES,
+ .formats = WM8961_FORMATS,},
+ .ops = &wm8961_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8961_dai);
+
+
+static struct snd_soc_codec *wm8961_codec;
+
+static int wm8961_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8961_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8961_codec;
+ codec = wm8961_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8961_snd_controls,
+ ARRAY_SIZE(wm8961_snd_controls));
+ snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
+ ARRAY_SIZE(wm8961_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8961_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8961_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8961_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 *reg_cache = codec->reg_cache;
+ int i;
+
+ for (i = 0; i < codec->reg_cache_size; i++) {
+ if (i == WM8961_SOFTWARE_RESET)
+ continue;
+
+ snd_soc_write(codec, i, reg_cache[i]);
+ }
+
+ wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define wm8961_suspend NULL
+#define wm8961_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_wm8961 = {
+ .probe = wm8961_probe,
+ .remove = wm8961_remove,
+ .suspend = wm8961_suspend,
+ .resume = wm8961_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8961);
+
+static int wm8961_register(struct wm8961_priv *wm8961)
+{
+ struct snd_soc_codec *codec = &wm8961->codec;
+ int ret;
+ u16 reg;
+
+ if (wm8961_codec) {
+ dev_err(codec->dev, "Another WM8961 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8961;
+ codec->name = "WM8961";
+ codec->owner = THIS_MODULE;
+ codec->dai = &wm8961_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8961->reg_cache);
+ codec->reg_cache = &wm8961->reg_cache;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8961_set_bias_level;
+ codec->volatile_register = wm8961_volatile_register;
+
+ memcpy(codec->reg_cache, wm8961_reg_defaults,
+ sizeof(wm8961_reg_defaults));
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ reg = snd_soc_read(codec, WM8961_SOFTWARE_RESET);
+ if (reg != 0x1801) {
+ dev_err(codec->dev, "Device is not a WM8961: ID=0x%x\n", reg);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ /* This isn't volatile - readback doesn't correspond to write */
+ reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME);
+ dev_info(codec->dev, "WM8961 family %d revision %c\n",
+ (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT,
+ ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT)
+ + 'A');
+
+ ret = wm8961_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ /* Enable class W */
+ reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B);
+ reg |= WM8961_CP_DYN_PWR_MASK;
+ snd_soc_write(codec, WM8961_CHARGE_PUMP_B, reg);
+
+ /* Latch volume update bits (right channel only, we always
+ * write both out) and default ZC on. */
+ reg = snd_soc_read(codec, WM8961_ROUT1_VOLUME);
+ snd_soc_write(codec, WM8961_ROUT1_VOLUME,
+ reg | WM8961_LO1ZC | WM8961_OUT1VU);
+ snd_soc_write(codec, WM8961_LOUT1_VOLUME, reg | WM8961_LO1ZC);
+ reg = snd_soc_read(codec, WM8961_ROUT2_VOLUME);
+ snd_soc_write(codec, WM8961_ROUT2_VOLUME,
+ reg | WM8961_SPKRZC | WM8961_SPKVU);
+ snd_soc_write(codec, WM8961_LOUT2_VOLUME, reg | WM8961_SPKLZC);
+
+ reg = snd_soc_read(codec, WM8961_RIGHT_ADC_VOLUME);
+ snd_soc_write(codec, WM8961_RIGHT_ADC_VOLUME, reg | WM8961_ADCVU);
+ reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME);
+ snd_soc_write(codec, WM8961_RIGHT_INPUT_VOLUME, reg | WM8961_IPVU);
+
+ /* Use soft mute by default */
+ reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2);
+ reg |= WM8961_DACSMM;
+ snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
+
+ /* Use automatic clocking mode by default; for now this is all
+ * we support.
+ */
+ reg = snd_soc_read(codec, WM8961_CLOCKING_3);
+ reg &= ~WM8961_MANUAL_MODE;
+ snd_soc_write(codec, WM8961_CLOCKING_3, reg);
+
+ wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ wm8961_dai.dev = codec->dev;
+
+ wm8961_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8961_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+
+err:
+ kfree(wm8961);
+ return ret;
+}
+
+static void wm8961_unregister(struct wm8961_priv *wm8961)
+{
+ wm8961_set_bias_level(&wm8961->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8961_dai);
+ snd_soc_unregister_codec(&wm8961->codec);
+ kfree(wm8961);
+ wm8961_codec = NULL;
+}
+
+static __devinit int wm8961_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8961_priv *wm8961;
+ struct snd_soc_codec *codec;
+
+ wm8961 = kzalloc(sizeof(struct wm8961_priv), GFP_KERNEL);
+ if (wm8961 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8961->codec;
+
+ i2c_set_clientdata(i2c, wm8961);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8961_register(wm8961);
+}
+
+static __devexit int wm8961_i2c_remove(struct i2c_client *client)
+{
+ struct wm8961_priv *wm8961 = i2c_get_clientdata(client);
+ wm8961_unregister(wm8961);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8961_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8961_i2c_suspend NULL
+#define wm8961_i2c_resume NULL
+#endif
+
+static const struct i2c_device_id wm8961_i2c_id[] = {
+ { "wm8961", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id);
+
+static struct i2c_driver wm8961_i2c_driver = {
+ .driver = {
+ .name = "wm8961",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8961_i2c_probe,
+ .remove = __devexit_p(wm8961_i2c_remove),
+ .suspend = wm8961_i2c_suspend,
+ .resume = wm8961_i2c_resume,
+ .id_table = wm8961_i2c_id,
+};
+
+static int __init wm8961_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8961_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8961 I2C driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(wm8961_modinit);
+
+static void __exit wm8961_exit(void)
+{
+ i2c_del_driver(&wm8961_i2c_driver);
+}
+module_exit(wm8961_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8961 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8961.h b/sound/soc/codecs/wm8961.h
new file mode 100644
index 0000000..5513bfd7
--- /dev/null
+++ b/sound/soc/codecs/wm8961.h
@@ -0,0 +1,866 @@
+/*
+ * wm8961.h -- WM8961 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8961_H
+#define _WM8961_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_codec_device soc_codec_dev_wm8961;
+extern struct snd_soc_dai wm8961_dai;
+
+#define WM8961_BCLK 1
+#define WM8961_LRCLK 2
+
+#define WM8961_BCLK_DIV_1 0
+#define WM8961_BCLK_DIV_1_5 1
+#define WM8961_BCLK_DIV_2 2
+#define WM8961_BCLK_DIV_3 3
+#define WM8961_BCLK_DIV_4 4
+#define WM8961_BCLK_DIV_5_5 5
+#define WM8961_BCLK_DIV_6 6
+#define WM8961_BCLK_DIV_8 7
+#define WM8961_BCLK_DIV_11 8
+#define WM8961_BCLK_DIV_12 9
+#define WM8961_BCLK_DIV_16 10
+#define WM8961_BCLK_DIV_24 11
+#define WM8961_BCLK_DIV_32 13
+
+
+/*
+ * Register values.
+ */
+#define WM8961_LEFT_INPUT_VOLUME 0x00
+#define WM8961_RIGHT_INPUT_VOLUME 0x01
+#define WM8961_LOUT1_VOLUME 0x02
+#define WM8961_ROUT1_VOLUME 0x03
+#define WM8961_CLOCKING1 0x04
+#define WM8961_ADC_DAC_CONTROL_1 0x05
+#define WM8961_ADC_DAC_CONTROL_2 0x06
+#define WM8961_AUDIO_INTERFACE_0 0x07
+#define WM8961_CLOCKING2 0x08
+#define WM8961_AUDIO_INTERFACE_1 0x09
+#define WM8961_LEFT_DAC_VOLUME 0x0A
+#define WM8961_RIGHT_DAC_VOLUME 0x0B
+#define WM8961_AUDIO_INTERFACE_2 0x0E
+#define WM8961_SOFTWARE_RESET 0x0F
+#define WM8961_ALC1 0x11
+#define WM8961_ALC2 0x12
+#define WM8961_ALC3 0x13
+#define WM8961_NOISE_GATE 0x14
+#define WM8961_LEFT_ADC_VOLUME 0x15
+#define WM8961_RIGHT_ADC_VOLUME 0x16
+#define WM8961_ADDITIONAL_CONTROL_1 0x17
+#define WM8961_ADDITIONAL_CONTROL_2 0x18
+#define WM8961_PWR_MGMT_1 0x19
+#define WM8961_PWR_MGMT_2 0x1A
+#define WM8961_ADDITIONAL_CONTROL_3 0x1B
+#define WM8961_ANTI_POP 0x1C
+#define WM8961_CLOCKING_3 0x1E
+#define WM8961_ADCL_SIGNAL_PATH 0x20
+#define WM8961_ADCR_SIGNAL_PATH 0x21
+#define WM8961_LOUT2_VOLUME 0x28
+#define WM8961_ROUT2_VOLUME 0x29
+#define WM8961_PWR_MGMT_3 0x2F
+#define WM8961_ADDITIONAL_CONTROL_4 0x30
+#define WM8961_CLASS_D_CONTROL_1 0x31
+#define WM8961_CLASS_D_CONTROL_2 0x33
+#define WM8961_CLOCKING_4 0x38
+#define WM8961_DSP_SIDETONE_0 0x39
+#define WM8961_DSP_SIDETONE_1 0x3A
+#define WM8961_DC_SERVO_0 0x3C
+#define WM8961_DC_SERVO_1 0x3D
+#define WM8961_DC_SERVO_3 0x3F
+#define WM8961_DC_SERVO_5 0x41
+#define WM8961_ANALOGUE_PGA_BIAS 0x44
+#define WM8961_ANALOGUE_HP_0 0x45
+#define WM8961_ANALOGUE_HP_2 0x47
+#define WM8961_CHARGE_PUMP_1 0x48
+#define WM8961_CHARGE_PUMP_B 0x52
+#define WM8961_WRITE_SEQUENCER_1 0x57
+#define WM8961_WRITE_SEQUENCER_2 0x58
+#define WM8961_WRITE_SEQUENCER_3 0x59
+#define WM8961_WRITE_SEQUENCER_4 0x5A
+#define WM8961_WRITE_SEQUENCER_5 0x5B
+#define WM8961_WRITE_SEQUENCER_6 0x5C
+#define WM8961_WRITE_SEQUENCER_7 0x5D
+#define WM8961_GENERAL_TEST_1 0xFC
+
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Left Input volume
+ */
+#define WM8961_IPVU 0x0100 /* IPVU */
+#define WM8961_IPVU_MASK 0x0100 /* IPVU */
+#define WM8961_IPVU_SHIFT 8 /* IPVU */
+#define WM8961_IPVU_WIDTH 1 /* IPVU */
+#define WM8961_LINMUTE 0x0080 /* LINMUTE */
+#define WM8961_LINMUTE_MASK 0x0080 /* LINMUTE */
+#define WM8961_LINMUTE_SHIFT 7 /* LINMUTE */
+#define WM8961_LINMUTE_WIDTH 1 /* LINMUTE */
+#define WM8961_LIZC 0x0040 /* LIZC */
+#define WM8961_LIZC_MASK 0x0040 /* LIZC */
+#define WM8961_LIZC_SHIFT 6 /* LIZC */
+#define WM8961_LIZC_WIDTH 1 /* LIZC */
+#define WM8961_LINVOL_MASK 0x003F /* LINVOL - [5:0] */
+#define WM8961_LINVOL_SHIFT 0 /* LINVOL - [5:0] */
+#define WM8961_LINVOL_WIDTH 6 /* LINVOL - [5:0] */
+
+/*
+ * R1 (0x01) - Right Input volume
+ */
+#define WM8961_DEVICE_ID_MASK 0xF000 /* DEVICE_ID - [15:12] */
+#define WM8961_DEVICE_ID_SHIFT 12 /* DEVICE_ID - [15:12] */
+#define WM8961_DEVICE_ID_WIDTH 4 /* DEVICE_ID - [15:12] */
+#define WM8961_CHIP_REV_MASK 0x0E00 /* CHIP_REV - [11:9] */
+#define WM8961_CHIP_REV_SHIFT 9 /* CHIP_REV - [11:9] */
+#define WM8961_CHIP_REV_WIDTH 3 /* CHIP_REV - [11:9] */
+#define WM8961_IPVU 0x0100 /* IPVU */
+#define WM8961_IPVU_MASK 0x0100 /* IPVU */
+#define WM8961_IPVU_SHIFT 8 /* IPVU */
+#define WM8961_IPVU_WIDTH 1 /* IPVU */
+#define WM8961_RINMUTE 0x0080 /* RINMUTE */
+#define WM8961_RINMUTE_MASK 0x0080 /* RINMUTE */
+#define WM8961_RINMUTE_SHIFT 7 /* RINMUTE */
+#define WM8961_RINMUTE_WIDTH 1 /* RINMUTE */
+#define WM8961_RIZC 0x0040 /* RIZC */
+#define WM8961_RIZC_MASK 0x0040 /* RIZC */
+#define WM8961_RIZC_SHIFT 6 /* RIZC */
+#define WM8961_RIZC_WIDTH 1 /* RIZC */
+#define WM8961_RINVOL_MASK 0x003F /* RINVOL - [5:0] */
+#define WM8961_RINVOL_SHIFT 0 /* RINVOL - [5:0] */
+#define WM8961_RINVOL_WIDTH 6 /* RINVOL - [5:0] */
+
+/*
+ * R2 (0x02) - LOUT1 volume
+ */
+#define WM8961_OUT1VU 0x0100 /* OUT1VU */
+#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */
+#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */
+#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */
+#define WM8961_LO1ZC 0x0080 /* LO1ZC */
+#define WM8961_LO1ZC_MASK 0x0080 /* LO1ZC */
+#define WM8961_LO1ZC_SHIFT 7 /* LO1ZC */
+#define WM8961_LO1ZC_WIDTH 1 /* LO1ZC */
+#define WM8961_LOUT1VOL_MASK 0x007F /* LOUT1VOL - [6:0] */
+#define WM8961_LOUT1VOL_SHIFT 0 /* LOUT1VOL - [6:0] */
+#define WM8961_LOUT1VOL_WIDTH 7 /* LOUT1VOL - [6:0] */
+
+/*
+ * R3 (0x03) - ROUT1 volume
+ */
+#define WM8961_OUT1VU 0x0100 /* OUT1VU */
+#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */
+#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */
+#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */
+#define WM8961_RO1ZC 0x0080 /* RO1ZC */
+#define WM8961_RO1ZC_MASK 0x0080 /* RO1ZC */
+#define WM8961_RO1ZC_SHIFT 7 /* RO1ZC */
+#define WM8961_RO1ZC_WIDTH 1 /* RO1ZC */
+#define WM8961_ROUT1VOL_MASK 0x007F /* ROUT1VOL - [6:0] */
+#define WM8961_ROUT1VOL_SHIFT 0 /* ROUT1VOL - [6:0] */
+#define WM8961_ROUT1VOL_WIDTH 7 /* ROUT1VOL - [6:0] */
+
+/*
+ * R4 (0x04) - Clocking1
+ */
+#define WM8961_ADCDIV_MASK 0x01C0 /* ADCDIV - [8:6] */
+#define WM8961_ADCDIV_SHIFT 6 /* ADCDIV - [8:6] */
+#define WM8961_ADCDIV_WIDTH 3 /* ADCDIV - [8:6] */
+#define WM8961_DACDIV_MASK 0x0038 /* DACDIV - [5:3] */
+#define WM8961_DACDIV_SHIFT 3 /* DACDIV - [5:3] */
+#define WM8961_DACDIV_WIDTH 3 /* DACDIV - [5:3] */
+#define WM8961_MCLKDIV 0x0004 /* MCLKDIV */
+#define WM8961_MCLKDIV_MASK 0x0004 /* MCLKDIV */
+#define WM8961_MCLKDIV_SHIFT 2 /* MCLKDIV */
+#define WM8961_MCLKDIV_WIDTH 1 /* MCLKDIV */
+
+/*
+ * R5 (0x05) - ADC & DAC Control 1
+ */
+#define WM8961_ADCPOL_MASK 0x0060 /* ADCPOL - [6:5] */
+#define WM8961_ADCPOL_SHIFT 5 /* ADCPOL - [6:5] */
+#define WM8961_ADCPOL_WIDTH 2 /* ADCPOL - [6:5] */
+#define WM8961_DACMU 0x0008 /* DACMU */
+#define WM8961_DACMU_MASK 0x0008 /* DACMU */
+#define WM8961_DACMU_SHIFT 3 /* DACMU */
+#define WM8961_DACMU_WIDTH 1 /* DACMU */
+#define WM8961_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */
+#define WM8961_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */
+#define WM8961_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */
+#define WM8961_ADCHPD 0x0001 /* ADCHPD */
+#define WM8961_ADCHPD_MASK 0x0001 /* ADCHPD */
+#define WM8961_ADCHPD_SHIFT 0 /* ADCHPD */
+#define WM8961_ADCHPD_WIDTH 1 /* ADCHPD */
+
+/*
+ * R6 (0x06) - ADC & DAC Control 2
+ */
+#define WM8961_ADC_HPF_CUT_MASK 0x0180 /* ADC_HPF_CUT - [8:7] */
+#define WM8961_ADC_HPF_CUT_SHIFT 7 /* ADC_HPF_CUT - [8:7] */
+#define WM8961_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [8:7] */
+#define WM8961_DACPOL_MASK 0x0060 /* DACPOL - [6:5] */
+#define WM8961_DACPOL_SHIFT 5 /* DACPOL - [6:5] */
+#define WM8961_DACPOL_WIDTH 2 /* DACPOL - [6:5] */
+#define WM8961_DACSMM 0x0008 /* DACSMM */
+#define WM8961_DACSMM_MASK 0x0008 /* DACSMM */
+#define WM8961_DACSMM_SHIFT 3 /* DACSMM */
+#define WM8961_DACSMM_WIDTH 1 /* DACSMM */
+#define WM8961_DACMR 0x0004 /* DACMR */
+#define WM8961_DACMR_MASK 0x0004 /* DACMR */
+#define WM8961_DACMR_SHIFT 2 /* DACMR */
+#define WM8961_DACMR_WIDTH 1 /* DACMR */
+#define WM8961_DACSLOPE 0x0002 /* DACSLOPE */
+#define WM8961_DACSLOPE_MASK 0x0002 /* DACSLOPE */
+#define WM8961_DACSLOPE_SHIFT 1 /* DACSLOPE */
+#define WM8961_DACSLOPE_WIDTH 1 /* DACSLOPE */
+#define WM8961_DAC_OSR128 0x0001 /* DAC_OSR128 */
+#define WM8961_DAC_OSR128_MASK 0x0001 /* DAC_OSR128 */
+#define WM8961_DAC_OSR128_SHIFT 0 /* DAC_OSR128 */
+#define WM8961_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */
+
+/*
+ * R7 (0x07) - Audio Interface 0
+ */
+#define WM8961_ALRSWAP 0x0100 /* ALRSWAP */
+#define WM8961_ALRSWAP_MASK 0x0100 /* ALRSWAP */
+#define WM8961_ALRSWAP_SHIFT 8 /* ALRSWAP */
+#define WM8961_ALRSWAP_WIDTH 1 /* ALRSWAP */
+#define WM8961_BCLKINV 0x0080 /* BCLKINV */
+#define WM8961_BCLKINV_MASK 0x0080 /* BCLKINV */
+#define WM8961_BCLKINV_SHIFT 7 /* BCLKINV */
+#define WM8961_BCLKINV_WIDTH 1 /* BCLKINV */
+#define WM8961_MS 0x0040 /* MS */
+#define WM8961_MS_MASK 0x0040 /* MS */
+#define WM8961_MS_SHIFT 6 /* MS */
+#define WM8961_MS_WIDTH 1 /* MS */
+#define WM8961_DLRSWAP 0x0020 /* DLRSWAP */
+#define WM8961_DLRSWAP_MASK 0x0020 /* DLRSWAP */
+#define WM8961_DLRSWAP_SHIFT 5 /* DLRSWAP */
+#define WM8961_DLRSWAP_WIDTH 1 /* DLRSWAP */
+#define WM8961_LRP 0x0010 /* LRP */
+#define WM8961_LRP_MASK 0x0010 /* LRP */
+#define WM8961_LRP_SHIFT 4 /* LRP */
+#define WM8961_LRP_WIDTH 1 /* LRP */
+#define WM8961_WL_MASK 0x000C /* WL - [3:2] */
+#define WM8961_WL_SHIFT 2 /* WL - [3:2] */
+#define WM8961_WL_WIDTH 2 /* WL - [3:2] */
+#define WM8961_FORMAT_MASK 0x0003 /* FORMAT - [1:0] */
+#define WM8961_FORMAT_SHIFT 0 /* FORMAT - [1:0] */
+#define WM8961_FORMAT_WIDTH 2 /* FORMAT - [1:0] */
+
+/*
+ * R8 (0x08) - Clocking2
+ */
+#define WM8961_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */
+#define WM8961_DCLKDIV_SHIFT 6 /* DCLKDIV - [8:6] */
+#define WM8961_DCLKDIV_WIDTH 3 /* DCLKDIV - [8:6] */
+#define WM8961_CLK_SYS_ENA 0x0020 /* CLK_SYS_ENA */
+#define WM8961_CLK_SYS_ENA_MASK 0x0020 /* CLK_SYS_ENA */
+#define WM8961_CLK_SYS_ENA_SHIFT 5 /* CLK_SYS_ENA */
+#define WM8961_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */
+#define WM8961_CLK_DSP_ENA 0x0010 /* CLK_DSP_ENA */
+#define WM8961_CLK_DSP_ENA_MASK 0x0010 /* CLK_DSP_ENA */
+#define WM8961_CLK_DSP_ENA_SHIFT 4 /* CLK_DSP_ENA */
+#define WM8961_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */
+#define WM8961_BCLKDIV_MASK 0x000F /* BCLKDIV - [3:0] */
+#define WM8961_BCLKDIV_SHIFT 0 /* BCLKDIV - [3:0] */
+#define WM8961_BCLKDIV_WIDTH 4 /* BCLKDIV - [3:0] */
+
+/*
+ * R9 (0x09) - Audio Interface 1
+ */
+#define WM8961_DACCOMP_MASK 0x0018 /* DACCOMP - [4:3] */
+#define WM8961_DACCOMP_SHIFT 3 /* DACCOMP - [4:3] */
+#define WM8961_DACCOMP_WIDTH 2 /* DACCOMP - [4:3] */
+#define WM8961_ADCCOMP_MASK 0x0006 /* ADCCOMP - [2:1] */
+#define WM8961_ADCCOMP_SHIFT 1 /* ADCCOMP - [2:1] */
+#define WM8961_ADCCOMP_WIDTH 2 /* ADCCOMP - [2:1] */
+#define WM8961_LOOPBACK 0x0001 /* LOOPBACK */
+#define WM8961_LOOPBACK_MASK 0x0001 /* LOOPBACK */
+#define WM8961_LOOPBACK_SHIFT 0 /* LOOPBACK */
+#define WM8961_LOOPBACK_WIDTH 1 /* LOOPBACK */
+
+/*
+ * R10 (0x0A) - Left DAC volume
+ */
+#define WM8961_DACVU 0x0100 /* DACVU */
+#define WM8961_DACVU_MASK 0x0100 /* DACVU */
+#define WM8961_DACVU_SHIFT 8 /* DACVU */
+#define WM8961_DACVU_WIDTH 1 /* DACVU */
+#define WM8961_LDACVOL_MASK 0x00FF /* LDACVOL - [7:0] */
+#define WM8961_LDACVOL_SHIFT 0 /* LDACVOL - [7:0] */
+#define WM8961_LDACVOL_WIDTH 8 /* LDACVOL - [7:0] */
+
+/*
+ * R11 (0x0B) - Right DAC volume
+ */
+#define WM8961_DACVU 0x0100 /* DACVU */
+#define WM8961_DACVU_MASK 0x0100 /* DACVU */
+#define WM8961_DACVU_SHIFT 8 /* DACVU */
+#define WM8961_DACVU_WIDTH 1 /* DACVU */
+#define WM8961_RDACVOL_MASK 0x00FF /* RDACVOL - [7:0] */
+#define WM8961_RDACVOL_SHIFT 0 /* RDACVOL - [7:0] */
+#define WM8961_RDACVOL_WIDTH 8 /* RDACVOL - [7:0] */
+
+/*
+ * R14 (0x0E) - Audio Interface 2
+ */
+#define WM8961_LRCLK_RATE_MASK 0x01FF /* LRCLK_RATE - [8:0] */
+#define WM8961_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [8:0] */
+#define WM8961_LRCLK_RATE_WIDTH 9 /* LRCLK_RATE - [8:0] */
+
+/*
+ * R15 (0x0F) - Software Reset
+ */
+#define WM8961_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */
+#define WM8961_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */
+#define WM8961_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */
+
+/*
+ * R17 (0x11) - ALC1
+ */
+#define WM8961_ALCSEL_MASK 0x0180 /* ALCSEL - [8:7] */
+#define WM8961_ALCSEL_SHIFT 7 /* ALCSEL - [8:7] */
+#define WM8961_ALCSEL_WIDTH 2 /* ALCSEL - [8:7] */
+#define WM8961_MAXGAIN_MASK 0x0070 /* MAXGAIN - [6:4] */
+#define WM8961_MAXGAIN_SHIFT 4 /* MAXGAIN - [6:4] */
+#define WM8961_MAXGAIN_WIDTH 3 /* MAXGAIN - [6:4] */
+#define WM8961_ALCL_MASK 0x000F /* ALCL - [3:0] */
+#define WM8961_ALCL_SHIFT 0 /* ALCL - [3:0] */
+#define WM8961_ALCL_WIDTH 4 /* ALCL - [3:0] */
+
+/*
+ * R18 (0x12) - ALC2
+ */
+#define WM8961_ALCZC 0x0080 /* ALCZC */
+#define WM8961_ALCZC_MASK 0x0080 /* ALCZC */
+#define WM8961_ALCZC_SHIFT 7 /* ALCZC */
+#define WM8961_ALCZC_WIDTH 1 /* ALCZC */
+#define WM8961_MINGAIN_MASK 0x0070 /* MINGAIN - [6:4] */
+#define WM8961_MINGAIN_SHIFT 4 /* MINGAIN - [6:4] */
+#define WM8961_MINGAIN_WIDTH 3 /* MINGAIN - [6:4] */
+#define WM8961_HLD_MASK 0x000F /* HLD - [3:0] */
+#define WM8961_HLD_SHIFT 0 /* HLD - [3:0] */
+#define WM8961_HLD_WIDTH 4 /* HLD - [3:0] */
+
+/*
+ * R19 (0x13) - ALC3
+ */
+#define WM8961_ALCMODE 0x0100 /* ALCMODE */
+#define WM8961_ALCMODE_MASK 0x0100 /* ALCMODE */
+#define WM8961_ALCMODE_SHIFT 8 /* ALCMODE */
+#define WM8961_ALCMODE_WIDTH 1 /* ALCMODE */
+#define WM8961_DCY_MASK 0x00F0 /* DCY - [7:4] */
+#define WM8961_DCY_SHIFT 4 /* DCY - [7:4] */
+#define WM8961_DCY_WIDTH 4 /* DCY - [7:4] */
+#define WM8961_ATK_MASK 0x000F /* ATK - [3:0] */
+#define WM8961_ATK_SHIFT 0 /* ATK - [3:0] */
+#define WM8961_ATK_WIDTH 4 /* ATK - [3:0] */
+
+/*
+ * R20 (0x14) - Noise Gate
+ */
+#define WM8961_NGTH_MASK 0x00F8 /* NGTH - [7:3] */
+#define WM8961_NGTH_SHIFT 3 /* NGTH - [7:3] */
+#define WM8961_NGTH_WIDTH 5 /* NGTH - [7:3] */
+#define WM8961_NGG 0x0002 /* NGG */
+#define WM8961_NGG_MASK 0x0002 /* NGG */
+#define WM8961_NGG_SHIFT 1 /* NGG */
+#define WM8961_NGG_WIDTH 1 /* NGG */
+#define WM8961_NGAT 0x0001 /* NGAT */
+#define WM8961_NGAT_MASK 0x0001 /* NGAT */
+#define WM8961_NGAT_SHIFT 0 /* NGAT */
+#define WM8961_NGAT_WIDTH 1 /* NGAT */
+
+/*
+ * R21 (0x15) - Left ADC volume
+ */
+#define WM8961_ADCVU 0x0100 /* ADCVU */
+#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8961_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8961_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8961_LADCVOL_MASK 0x00FF /* LADCVOL - [7:0] */
+#define WM8961_LADCVOL_SHIFT 0 /* LADCVOL - [7:0] */
+#define WM8961_LADCVOL_WIDTH 8 /* LADCVOL - [7:0] */
+
+/*
+ * R22 (0x16) - Right ADC volume
+ */
+#define WM8961_ADCVU 0x0100 /* ADCVU */
+#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8961_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8961_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8961_RADCVOL_MASK 0x00FF /* RADCVOL - [7:0] */
+#define WM8961_RADCVOL_SHIFT 0 /* RADCVOL - [7:0] */
+#define WM8961_RADCVOL_WIDTH 8 /* RADCVOL - [7:0] */
+
+/*
+ * R23 (0x17) - Additional control(1)
+ */
+#define WM8961_TSDEN 0x0100 /* TSDEN */
+#define WM8961_TSDEN_MASK 0x0100 /* TSDEN */
+#define WM8961_TSDEN_SHIFT 8 /* TSDEN */
+#define WM8961_TSDEN_WIDTH 1 /* TSDEN */
+#define WM8961_DMONOMIX 0x0010 /* DMONOMIX */
+#define WM8961_DMONOMIX_MASK 0x0010 /* DMONOMIX */
+#define WM8961_DMONOMIX_SHIFT 4 /* DMONOMIX */
+#define WM8961_DMONOMIX_WIDTH 1 /* DMONOMIX */
+#define WM8961_TOEN 0x0001 /* TOEN */
+#define WM8961_TOEN_MASK 0x0001 /* TOEN */
+#define WM8961_TOEN_SHIFT 0 /* TOEN */
+#define WM8961_TOEN_WIDTH 1 /* TOEN */
+
+/*
+ * R24 (0x18) - Additional control(2)
+ */
+#define WM8961_TRIS 0x0008 /* TRIS */
+#define WM8961_TRIS_MASK 0x0008 /* TRIS */
+#define WM8961_TRIS_SHIFT 3 /* TRIS */
+#define WM8961_TRIS_WIDTH 1 /* TRIS */
+
+/*
+ * R25 (0x19) - Pwr Mgmt (1)
+ */
+#define WM8961_VMIDSEL_MASK 0x0180 /* VMIDSEL - [8:7] */
+#define WM8961_VMIDSEL_SHIFT 7 /* VMIDSEL - [8:7] */
+#define WM8961_VMIDSEL_WIDTH 2 /* VMIDSEL - [8:7] */
+#define WM8961_VREF 0x0040 /* VREF */
+#define WM8961_VREF_MASK 0x0040 /* VREF */
+#define WM8961_VREF_SHIFT 6 /* VREF */
+#define WM8961_VREF_WIDTH 1 /* VREF */
+#define WM8961_AINL 0x0020 /* AINL */
+#define WM8961_AINL_MASK 0x0020 /* AINL */
+#define WM8961_AINL_SHIFT 5 /* AINL */
+#define WM8961_AINL_WIDTH 1 /* AINL */
+#define WM8961_AINR 0x0010 /* AINR */
+#define WM8961_AINR_MASK 0x0010 /* AINR */
+#define WM8961_AINR_SHIFT 4 /* AINR */
+#define WM8961_AINR_WIDTH 1 /* AINR */
+#define WM8961_ADCL 0x0008 /* ADCL */
+#define WM8961_ADCL_MASK 0x0008 /* ADCL */
+#define WM8961_ADCL_SHIFT 3 /* ADCL */
+#define WM8961_ADCL_WIDTH 1 /* ADCL */
+#define WM8961_ADCR 0x0004 /* ADCR */
+#define WM8961_ADCR_MASK 0x0004 /* ADCR */
+#define WM8961_ADCR_SHIFT 2 /* ADCR */
+#define WM8961_ADCR_WIDTH 1 /* ADCR */
+#define WM8961_MICB 0x0002 /* MICB */
+#define WM8961_MICB_MASK 0x0002 /* MICB */
+#define WM8961_MICB_SHIFT 1 /* MICB */
+#define WM8961_MICB_WIDTH 1 /* MICB */
+
+/*
+ * R26 (0x1A) - Pwr Mgmt (2)
+ */
+#define WM8961_DACL 0x0100 /* DACL */
+#define WM8961_DACL_MASK 0x0100 /* DACL */
+#define WM8961_DACL_SHIFT 8 /* DACL */
+#define WM8961_DACL_WIDTH 1 /* DACL */
+#define WM8961_DACR 0x0080 /* DACR */
+#define WM8961_DACR_MASK 0x0080 /* DACR */
+#define WM8961_DACR_SHIFT 7 /* DACR */
+#define WM8961_DACR_WIDTH 1 /* DACR */
+#define WM8961_LOUT1_PGA 0x0040 /* LOUT1_PGA */
+#define WM8961_LOUT1_PGA_MASK 0x0040 /* LOUT1_PGA */
+#define WM8961_LOUT1_PGA_SHIFT 6 /* LOUT1_PGA */
+#define WM8961_LOUT1_PGA_WIDTH 1 /* LOUT1_PGA */
+#define WM8961_ROUT1_PGA 0x0020 /* ROUT1_PGA */
+#define WM8961_ROUT1_PGA_MASK 0x0020 /* ROUT1_PGA */
+#define WM8961_ROUT1_PGA_SHIFT 5 /* ROUT1_PGA */
+#define WM8961_ROUT1_PGA_WIDTH 1 /* ROUT1_PGA */
+#define WM8961_SPKL_PGA 0x0010 /* SPKL_PGA */
+#define WM8961_SPKL_PGA_MASK 0x0010 /* SPKL_PGA */
+#define WM8961_SPKL_PGA_SHIFT 4 /* SPKL_PGA */
+#define WM8961_SPKL_PGA_WIDTH 1 /* SPKL_PGA */
+#define WM8961_SPKR_PGA 0x0008 /* SPKR_PGA */
+#define WM8961_SPKR_PGA_MASK 0x0008 /* SPKR_PGA */
+#define WM8961_SPKR_PGA_SHIFT 3 /* SPKR_PGA */
+#define WM8961_SPKR_PGA_WIDTH 1 /* SPKR_PGA */
+
+/*
+ * R27 (0x1B) - Additional Control (3)
+ */
+#define WM8961_SAMPLE_RATE_MASK 0x0007 /* SAMPLE_RATE - [2:0] */
+#define WM8961_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [2:0] */
+#define WM8961_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [2:0] */
+
+/*
+ * R28 (0x1C) - Anti-pop
+ */
+#define WM8961_BUFDCOPEN 0x0010 /* BUFDCOPEN */
+#define WM8961_BUFDCOPEN_MASK 0x0010 /* BUFDCOPEN */
+#define WM8961_BUFDCOPEN_SHIFT 4 /* BUFDCOPEN */
+#define WM8961_BUFDCOPEN_WIDTH 1 /* BUFDCOPEN */
+#define WM8961_BUFIOEN 0x0008 /* BUFIOEN */
+#define WM8961_BUFIOEN_MASK 0x0008 /* BUFIOEN */
+#define WM8961_BUFIOEN_SHIFT 3 /* BUFIOEN */
+#define WM8961_BUFIOEN_WIDTH 1 /* BUFIOEN */
+#define WM8961_SOFT_ST 0x0004 /* SOFT_ST */
+#define WM8961_SOFT_ST_MASK 0x0004 /* SOFT_ST */
+#define WM8961_SOFT_ST_SHIFT 2 /* SOFT_ST */
+#define WM8961_SOFT_ST_WIDTH 1 /* SOFT_ST */
+
+/*
+ * R30 (0x1E) - Clocking 3
+ */
+#define WM8961_CLK_TO_DIV_MASK 0x0180 /* CLK_TO_DIV - [8:7] */
+#define WM8961_CLK_TO_DIV_SHIFT 7 /* CLK_TO_DIV - [8:7] */
+#define WM8961_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [8:7] */
+#define WM8961_CLK_256K_DIV_MASK 0x007E /* CLK_256K_DIV - [6:1] */
+#define WM8961_CLK_256K_DIV_SHIFT 1 /* CLK_256K_DIV - [6:1] */
+#define WM8961_CLK_256K_DIV_WIDTH 6 /* CLK_256K_DIV - [6:1] */
+#define WM8961_MANUAL_MODE 0x0001 /* MANUAL_MODE */
+#define WM8961_MANUAL_MODE_MASK 0x0001 /* MANUAL_MODE */
+#define WM8961_MANUAL_MODE_SHIFT 0 /* MANUAL_MODE */
+#define WM8961_MANUAL_MODE_WIDTH 1 /* MANUAL_MODE */
+
+/*
+ * R32 (0x20) - ADCL signal path
+ */
+#define WM8961_LMICBOOST_MASK 0x0030 /* LMICBOOST - [5:4] */
+#define WM8961_LMICBOOST_SHIFT 4 /* LMICBOOST - [5:4] */
+#define WM8961_LMICBOOST_WIDTH 2 /* LMICBOOST - [5:4] */
+
+/*
+ * R33 (0x21) - ADCR signal path
+ */
+#define WM8961_RMICBOOST_MASK 0x0030 /* RMICBOOST - [5:4] */
+#define WM8961_RMICBOOST_SHIFT 4 /* RMICBOOST - [5:4] */
+#define WM8961_RMICBOOST_WIDTH 2 /* RMICBOOST - [5:4] */
+
+/*
+ * R40 (0x28) - LOUT2 volume
+ */
+#define WM8961_SPKVU 0x0100 /* SPKVU */
+#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */
+#define WM8961_SPKVU_SHIFT 8 /* SPKVU */
+#define WM8961_SPKVU_WIDTH 1 /* SPKVU */
+#define WM8961_SPKLZC 0x0080 /* SPKLZC */
+#define WM8961_SPKLZC_MASK 0x0080 /* SPKLZC */
+#define WM8961_SPKLZC_SHIFT 7 /* SPKLZC */
+#define WM8961_SPKLZC_WIDTH 1 /* SPKLZC */
+#define WM8961_SPKLVOL_MASK 0x007F /* SPKLVOL - [6:0] */
+#define WM8961_SPKLVOL_SHIFT 0 /* SPKLVOL - [6:0] */
+#define WM8961_SPKLVOL_WIDTH 7 /* SPKLVOL - [6:0] */
+
+/*
+ * R41 (0x29) - ROUT2 volume
+ */
+#define WM8961_SPKVU 0x0100 /* SPKVU */
+#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */
+#define WM8961_SPKVU_SHIFT 8 /* SPKVU */
+#define WM8961_SPKVU_WIDTH 1 /* SPKVU */
+#define WM8961_SPKRZC 0x0080 /* SPKRZC */
+#define WM8961_SPKRZC_MASK 0x0080 /* SPKRZC */
+#define WM8961_SPKRZC_SHIFT 7 /* SPKRZC */
+#define WM8961_SPKRZC_WIDTH 1 /* SPKRZC */
+#define WM8961_SPKRVOL_MASK 0x007F /* SPKRVOL - [6:0] */
+#define WM8961_SPKRVOL_SHIFT 0 /* SPKRVOL - [6:0] */
+#define WM8961_SPKRVOL_WIDTH 7 /* SPKRVOL - [6:0] */
+
+/*
+ * R47 (0x2F) - Pwr Mgmt (3)
+ */
+#define WM8961_TEMP_SHUT 0x0002 /* TEMP_SHUT */
+#define WM8961_TEMP_SHUT_MASK 0x0002 /* TEMP_SHUT */
+#define WM8961_TEMP_SHUT_SHIFT 1 /* TEMP_SHUT */
+#define WM8961_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */
+#define WM8961_TEMP_WARN 0x0001 /* TEMP_WARN */
+#define WM8961_TEMP_WARN_MASK 0x0001 /* TEMP_WARN */
+#define WM8961_TEMP_WARN_SHIFT 0 /* TEMP_WARN */
+#define WM8961_TEMP_WARN_WIDTH 1 /* TEMP_WARN */
+
+/*
+ * R48 (0x30) - Additional Control (4)
+ */
+#define WM8961_TSENSEN 0x0002 /* TSENSEN */
+#define WM8961_TSENSEN_MASK 0x0002 /* TSENSEN */
+#define WM8961_TSENSEN_SHIFT 1 /* TSENSEN */
+#define WM8961_TSENSEN_WIDTH 1 /* TSENSEN */
+#define WM8961_MBSEL 0x0001 /* MBSEL */
+#define WM8961_MBSEL_MASK 0x0001 /* MBSEL */
+#define WM8961_MBSEL_SHIFT 0 /* MBSEL */
+#define WM8961_MBSEL_WIDTH 1 /* MBSEL */
+
+/*
+ * R49 (0x31) - Class D Control 1
+ */
+#define WM8961_SPKR_ENA 0x0080 /* SPKR_ENA */
+#define WM8961_SPKR_ENA_MASK 0x0080 /* SPKR_ENA */
+#define WM8961_SPKR_ENA_SHIFT 7 /* SPKR_ENA */
+#define WM8961_SPKR_ENA_WIDTH 1 /* SPKR_ENA */
+#define WM8961_SPKL_ENA 0x0040 /* SPKL_ENA */
+#define WM8961_SPKL_ENA_MASK 0x0040 /* SPKL_ENA */
+#define WM8961_SPKL_ENA_SHIFT 6 /* SPKL_ENA */
+#define WM8961_SPKL_ENA_WIDTH 1 /* SPKL_ENA */
+
+/*
+ * R51 (0x33) - Class D Control 2
+ */
+#define WM8961_CLASSD_ACGAIN_MASK 0x0007 /* CLASSD_ACGAIN - [2:0] */
+#define WM8961_CLASSD_ACGAIN_SHIFT 0 /* CLASSD_ACGAIN - [2:0] */
+#define WM8961_CLASSD_ACGAIN_WIDTH 3 /* CLASSD_ACGAIN - [2:0] */
+
+/*
+ * R56 (0x38) - Clocking 4
+ */
+#define WM8961_CLK_DCS_DIV_MASK 0x01E0 /* CLK_DCS_DIV - [8:5] */
+#define WM8961_CLK_DCS_DIV_SHIFT 5 /* CLK_DCS_DIV - [8:5] */
+#define WM8961_CLK_DCS_DIV_WIDTH 4 /* CLK_DCS_DIV - [8:5] */
+#define WM8961_CLK_SYS_RATE_MASK 0x001E /* CLK_SYS_RATE - [4:1] */
+#define WM8961_CLK_SYS_RATE_SHIFT 1 /* CLK_SYS_RATE - [4:1] */
+#define WM8961_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [4:1] */
+
+/*
+ * R57 (0x39) - DSP Sidetone 0
+ */
+#define WM8961_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8961_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8961_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8961_ADC_TO_DACR_MASK 0x000C /* ADC_TO_DACR - [3:2] */
+#define WM8961_ADC_TO_DACR_SHIFT 2 /* ADC_TO_DACR - [3:2] */
+#define WM8961_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [3:2] */
+
+/*
+ * R58 (0x3A) - DSP Sidetone 1
+ */
+#define WM8961_ADCL_DAC_SVOL_MASK 0x00F0 /* ADCL_DAC_SVOL - [7:4] */
+#define WM8961_ADCL_DAC_SVOL_SHIFT 4 /* ADCL_DAC_SVOL - [7:4] */
+#define WM8961_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [7:4] */
+#define WM8961_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */
+#define WM8961_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */
+#define WM8961_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */
+
+/*
+ * R60 (0x3C) - DC Servo 0
+ */
+#define WM8961_DCS_ENA_CHAN_INL 0x0080 /* DCS_ENA_CHAN_INL */
+#define WM8961_DCS_ENA_CHAN_INL_MASK 0x0080 /* DCS_ENA_CHAN_INL */
+#define WM8961_DCS_ENA_CHAN_INL_SHIFT 7 /* DCS_ENA_CHAN_INL */
+#define WM8961_DCS_ENA_CHAN_INL_WIDTH 1 /* DCS_ENA_CHAN_INL */
+#define WM8961_DCS_TRIG_STARTUP_INL 0x0040 /* DCS_TRIG_STARTUP_INL */
+#define WM8961_DCS_TRIG_STARTUP_INL_MASK 0x0040 /* DCS_TRIG_STARTUP_INL */
+#define WM8961_DCS_TRIG_STARTUP_INL_SHIFT 6 /* DCS_TRIG_STARTUP_INL */
+#define WM8961_DCS_TRIG_STARTUP_INL_WIDTH 1 /* DCS_TRIG_STARTUP_INL */
+#define WM8961_DCS_TRIG_SERIES_INL 0x0010 /* DCS_TRIG_SERIES_INL */
+#define WM8961_DCS_TRIG_SERIES_INL_MASK 0x0010 /* DCS_TRIG_SERIES_INL */
+#define WM8961_DCS_TRIG_SERIES_INL_SHIFT 4 /* DCS_TRIG_SERIES_INL */
+#define WM8961_DCS_TRIG_SERIES_INL_WIDTH 1 /* DCS_TRIG_SERIES_INL */
+#define WM8961_DCS_ENA_CHAN_INR 0x0008 /* DCS_ENA_CHAN_INR */
+#define WM8961_DCS_ENA_CHAN_INR_MASK 0x0008 /* DCS_ENA_CHAN_INR */
+#define WM8961_DCS_ENA_CHAN_INR_SHIFT 3 /* DCS_ENA_CHAN_INR */
+#define WM8961_DCS_ENA_CHAN_INR_WIDTH 1 /* DCS_ENA_CHAN_INR */
+#define WM8961_DCS_TRIG_STARTUP_INR 0x0004 /* DCS_TRIG_STARTUP_INR */
+#define WM8961_DCS_TRIG_STARTUP_INR_MASK 0x0004 /* DCS_TRIG_STARTUP_INR */
+#define WM8961_DCS_TRIG_STARTUP_INR_SHIFT 2 /* DCS_TRIG_STARTUP_INR */
+#define WM8961_DCS_TRIG_STARTUP_INR_WIDTH 1 /* DCS_TRIG_STARTUP_INR */
+#define WM8961_DCS_TRIG_SERIES_INR 0x0001 /* DCS_TRIG_SERIES_INR */
+#define WM8961_DCS_TRIG_SERIES_INR_MASK 0x0001 /* DCS_TRIG_SERIES_INR */
+#define WM8961_DCS_TRIG_SERIES_INR_SHIFT 0 /* DCS_TRIG_SERIES_INR */
+#define WM8961_DCS_TRIG_SERIES_INR_WIDTH 1 /* DCS_TRIG_SERIES_INR */
+
+/*
+ * R61 (0x3D) - DC Servo 1
+ */
+#define WM8961_DCS_ENA_CHAN_HPL 0x0080 /* DCS_ENA_CHAN_HPL */
+#define WM8961_DCS_ENA_CHAN_HPL_MASK 0x0080 /* DCS_ENA_CHAN_HPL */
+#define WM8961_DCS_ENA_CHAN_HPL_SHIFT 7 /* DCS_ENA_CHAN_HPL */
+#define WM8961_DCS_ENA_CHAN_HPL_WIDTH 1 /* DCS_ENA_CHAN_HPL */
+#define WM8961_DCS_TRIG_STARTUP_HPL 0x0040 /* DCS_TRIG_STARTUP_HPL */
+#define WM8961_DCS_TRIG_STARTUP_HPL_MASK 0x0040 /* DCS_TRIG_STARTUP_HPL */
+#define WM8961_DCS_TRIG_STARTUP_HPL_SHIFT 6 /* DCS_TRIG_STARTUP_HPL */
+#define WM8961_DCS_TRIG_STARTUP_HPL_WIDTH 1 /* DCS_TRIG_STARTUP_HPL */
+#define WM8961_DCS_TRIG_SERIES_HPL 0x0010 /* DCS_TRIG_SERIES_HPL */
+#define WM8961_DCS_TRIG_SERIES_HPL_MASK 0x0010 /* DCS_TRIG_SERIES_HPL */
+#define WM8961_DCS_TRIG_SERIES_HPL_SHIFT 4 /* DCS_TRIG_SERIES_HPL */
+#define WM8961_DCS_TRIG_SERIES_HPL_WIDTH 1 /* DCS_TRIG_SERIES_HPL */
+#define WM8961_DCS_ENA_CHAN_HPR 0x0008 /* DCS_ENA_CHAN_HPR */
+#define WM8961_DCS_ENA_CHAN_HPR_MASK 0x0008 /* DCS_ENA_CHAN_HPR */
+#define WM8961_DCS_ENA_CHAN_HPR_SHIFT 3 /* DCS_ENA_CHAN_HPR */
+#define WM8961_DCS_ENA_CHAN_HPR_WIDTH 1 /* DCS_ENA_CHAN_HPR */
+#define WM8961_DCS_TRIG_STARTUP_HPR 0x0004 /* DCS_TRIG_STARTUP_HPR */
+#define WM8961_DCS_TRIG_STARTUP_HPR_MASK 0x0004 /* DCS_TRIG_STARTUP_HPR */
+#define WM8961_DCS_TRIG_STARTUP_HPR_SHIFT 2 /* DCS_TRIG_STARTUP_HPR */
+#define WM8961_DCS_TRIG_STARTUP_HPR_WIDTH 1 /* DCS_TRIG_STARTUP_HPR */
+#define WM8961_DCS_TRIG_SERIES_HPR 0x0001 /* DCS_TRIG_SERIES_HPR */
+#define WM8961_DCS_TRIG_SERIES_HPR_MASK 0x0001 /* DCS_TRIG_SERIES_HPR */
+#define WM8961_DCS_TRIG_SERIES_HPR_SHIFT 0 /* DCS_TRIG_SERIES_HPR */
+#define WM8961_DCS_TRIG_SERIES_HPR_WIDTH 1 /* DCS_TRIG_SERIES_HPR */
+
+/*
+ * R63 (0x3F) - DC Servo 3
+ */
+#define WM8961_DCS_FILT_BW_SERIES_MASK 0x0030 /* DCS_FILT_BW_SERIES - [5:4] */
+#define WM8961_DCS_FILT_BW_SERIES_SHIFT 4 /* DCS_FILT_BW_SERIES - [5:4] */
+#define WM8961_DCS_FILT_BW_SERIES_WIDTH 2 /* DCS_FILT_BW_SERIES - [5:4] */
+
+/*
+ * R65 (0x41) - DC Servo 5
+ */
+#define WM8961_DCS_SERIES_NO_HP_MASK 0x007F /* DCS_SERIES_NO_HP - [6:0] */
+#define WM8961_DCS_SERIES_NO_HP_SHIFT 0 /* DCS_SERIES_NO_HP - [6:0] */
+#define WM8961_DCS_SERIES_NO_HP_WIDTH 7 /* DCS_SERIES_NO_HP - [6:0] */
+
+/*
+ * R68 (0x44) - Analogue PGA Bias
+ */
+#define WM8961_HP_PGAS_BIAS_MASK 0x0007 /* HP_PGAS_BIAS - [2:0] */
+#define WM8961_HP_PGAS_BIAS_SHIFT 0 /* HP_PGAS_BIAS - [2:0] */
+#define WM8961_HP_PGAS_BIAS_WIDTH 3 /* HP_PGAS_BIAS - [2:0] */
+
+/*
+ * R69 (0x45) - Analogue HP 0
+ */
+#define WM8961_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */
+#define WM8961_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */
+#define WM8961_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */
+#define WM8961_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */
+#define WM8961_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */
+#define WM8961_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */
+#define WM8961_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */
+#define WM8961_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */
+#define WM8961_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */
+#define WM8961_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */
+#define WM8961_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */
+#define WM8961_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */
+#define WM8961_HPL_ENA 0x0010 /* HPL_ENA */
+#define WM8961_HPL_ENA_MASK 0x0010 /* HPL_ENA */
+#define WM8961_HPL_ENA_SHIFT 4 /* HPL_ENA */
+#define WM8961_HPL_ENA_WIDTH 1 /* HPL_ENA */
+#define WM8961_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */
+#define WM8961_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */
+#define WM8961_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */
+#define WM8961_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */
+#define WM8961_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */
+#define WM8961_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */
+#define WM8961_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */
+#define WM8961_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */
+#define WM8961_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */
+#define WM8961_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */
+#define WM8961_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */
+#define WM8961_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */
+#define WM8961_HPR_ENA 0x0001 /* HPR_ENA */
+#define WM8961_HPR_ENA_MASK 0x0001 /* HPR_ENA */
+#define WM8961_HPR_ENA_SHIFT 0 /* HPR_ENA */
+#define WM8961_HPR_ENA_WIDTH 1 /* HPR_ENA */
+
+/*
+ * R71 (0x47) - Analogue HP 2
+ */
+#define WM8961_HPL_VOL_MASK 0x01C0 /* HPL_VOL - [8:6] */
+#define WM8961_HPL_VOL_SHIFT 6 /* HPL_VOL - [8:6] */
+#define WM8961_HPL_VOL_WIDTH 3 /* HPL_VOL - [8:6] */
+#define WM8961_HPR_VOL_MASK 0x0038 /* HPR_VOL - [5:3] */
+#define WM8961_HPR_VOL_SHIFT 3 /* HPR_VOL - [5:3] */
+#define WM8961_HPR_VOL_WIDTH 3 /* HPR_VOL - [5:3] */
+#define WM8961_HP_BIAS_BOOST_MASK 0x0007 /* HP_BIAS_BOOST - [2:0] */
+#define WM8961_HP_BIAS_BOOST_SHIFT 0 /* HP_BIAS_BOOST - [2:0] */
+#define WM8961_HP_BIAS_BOOST_WIDTH 3 /* HP_BIAS_BOOST - [2:0] */
+
+/*
+ * R72 (0x48) - Charge Pump 1
+ */
+#define WM8961_CP_ENA 0x0001 /* CP_ENA */
+#define WM8961_CP_ENA_MASK 0x0001 /* CP_ENA */
+#define WM8961_CP_ENA_SHIFT 0 /* CP_ENA */
+#define WM8961_CP_ENA_WIDTH 1 /* CP_ENA */
+
+/*
+ * R82 (0x52) - Charge Pump B
+ */
+#define WM8961_CP_DYN_PWR_MASK 0x0003 /* CP_DYN_PWR - [1:0] */
+#define WM8961_CP_DYN_PWR_SHIFT 0 /* CP_DYN_PWR - [1:0] */
+#define WM8961_CP_DYN_PWR_WIDTH 2 /* CP_DYN_PWR - [1:0] */
+
+/*
+ * R87 (0x57) - Write Sequencer 1
+ */
+#define WM8961_WSEQ_ENA 0x0020 /* WSEQ_ENA */
+#define WM8961_WSEQ_ENA_MASK 0x0020 /* WSEQ_ENA */
+#define WM8961_WSEQ_ENA_SHIFT 5 /* WSEQ_ENA */
+#define WM8961_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */
+#define WM8961_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8961_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8961_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */
+
+/*
+ * R88 (0x58) - Write Sequencer 2
+ */
+#define WM8961_WSEQ_EOS 0x0100 /* WSEQ_EOS */
+#define WM8961_WSEQ_EOS_MASK 0x0100 /* WSEQ_EOS */
+#define WM8961_WSEQ_EOS_SHIFT 8 /* WSEQ_EOS */
+#define WM8961_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */
+#define WM8961_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */
+#define WM8961_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */
+#define WM8961_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */
+
+/*
+ * R89 (0x59) - Write Sequencer 3
+ */
+#define WM8961_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */
+#define WM8961_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */
+#define WM8961_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */
+
+/*
+ * R90 (0x5A) - Write Sequencer 4
+ */
+#define WM8961_WSEQ_ABORT 0x0100 /* WSEQ_ABORT */
+#define WM8961_WSEQ_ABORT_MASK 0x0100 /* WSEQ_ABORT */
+#define WM8961_WSEQ_ABORT_SHIFT 8 /* WSEQ_ABORT */
+#define WM8961_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */
+#define WM8961_WSEQ_START 0x0080 /* WSEQ_START */
+#define WM8961_WSEQ_START_MASK 0x0080 /* WSEQ_START */
+#define WM8961_WSEQ_START_SHIFT 7 /* WSEQ_START */
+#define WM8961_WSEQ_START_WIDTH 1 /* WSEQ_START */
+#define WM8961_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */
+#define WM8961_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */
+#define WM8961_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */
+
+/*
+ * R91 (0x5B) - Write Sequencer 5
+ */
+#define WM8961_WSEQ_DATA_WIDTH_MASK 0x0070 /* WSEQ_DATA_WIDTH - [6:4] */
+#define WM8961_WSEQ_DATA_WIDTH_SHIFT 4 /* WSEQ_DATA_WIDTH - [6:4] */
+#define WM8961_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [6:4] */
+#define WM8961_WSEQ_DATA_START_MASK 0x000F /* WSEQ_DATA_START - [3:0] */
+#define WM8961_WSEQ_DATA_START_SHIFT 0 /* WSEQ_DATA_START - [3:0] */
+#define WM8961_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [3:0] */
+
+/*
+ * R92 (0x5C) - Write Sequencer 6
+ */
+#define WM8961_WSEQ_DELAY_MASK 0x000F /* WSEQ_DELAY - [3:0] */
+#define WM8961_WSEQ_DELAY_SHIFT 0 /* WSEQ_DELAY - [3:0] */
+#define WM8961_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [3:0] */
+
+/*
+ * R93 (0x5D) - Write Sequencer 7
+ */
+#define WM8961_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */
+#define WM8961_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */
+#define WM8961_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */
+#define WM8961_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */
+
+/*
+ * R252 (0xFC) - General test 1
+ */
+#define WM8961_ARA_ENA 0x0002 /* ARA_ENA */
+#define WM8961_ARA_ENA_MASK 0x0002 /* ARA_ENA */
+#define WM8961_ARA_ENA_SHIFT 1 /* ARA_ENA */
+#define WM8961_ARA_ENA_WIDTH 1 /* ARA_ENA */
+#define WM8961_AUTO_INC 0x0001 /* AUTO_INC */
+#define WM8961_AUTO_INC_MASK 0x0001 /* AUTO_INC */
+#define WM8961_AUTO_INC_SHIFT 0 /* AUTO_INC */
+#define WM8961_AUTO_INC_WIDTH 1 /* AUTO_INC */
+
+#endif
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 032dca2..d66efb0 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -59,44 +59,7 @@ static const u16 wm8971_reg[] = {
0x0079, 0x0079, 0x0079, /* 40 */
};
-static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg < WM8971_REG_COUNT)
- return cache[reg];
-
- return -1;
-}
-
-static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg < WM8971_REG_COUNT)
- cache[reg] = value;
-}
-
-static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8753 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8971_write_reg_cache (codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0)
+#define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0)
/* WM8971 Controls */
static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" };
@@ -521,7 +484,7 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8971_write(codec, WM8971_IFACE, iface);
+ snd_soc_write(codec, WM8971_IFACE, iface);
return 0;
}
@@ -533,8 +496,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct wm8971_priv *wm8971 = codec->private_data;
- u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
- u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
+ u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3;
+ u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0;
int coeff = get_coeff(wm8971->sysclk, params_rate(params));
/* bit size */
@@ -553,9 +516,9 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
}
/* set iface & srate */
- wm8971_write(codec, WM8971_IFACE, iface);
+ snd_soc_write(codec, WM8971_IFACE, iface);
if (coeff >= 0)
- wm8971_write(codec, WM8971_SRATE, srate |
+ snd_soc_write(codec, WM8971_SRATE, srate |
(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
return 0;
@@ -564,33 +527,33 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
static int wm8971_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7;
+ u16 mute_reg = snd_soc_read(codec, WM8971_ADCDAC) & 0xfff7;
if (mute)
- wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8);
+ snd_soc_write(codec, WM8971_ADCDAC, mute_reg | 0x8);
else
- wm8971_write(codec, WM8971_ADCDAC, mute_reg);
+ snd_soc_write(codec, WM8971_ADCDAC, mute_reg);
return 0;
}
static int wm8971_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
+ u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
switch (level) {
case SND_SOC_BIAS_ON:
/* set vmid to 50k and unmute dac */
- wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1);
+ snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* mute dac and set vmid to 500k, enable VREF */
- wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
+ snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
break;
case SND_SOC_BIAS_OFF:
- wm8971_write(codec, WM8971_PWR1, 0x0001);
+ snd_soc_write(codec, WM8971_PWR1, 0x0001);
break;
}
codec->bias_level = level;
@@ -667,8 +630,8 @@ static int wm8971_resume(struct platform_device *pdev)
/* charge wm8971 caps */
if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
- reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
- wm8971_write(codec, WM8971_PWR1, reg | 0x01c0);
+ reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
+ snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
codec->bias_level = SND_SOC_BIAS_ON;
queue_delayed_work(wm8971_workq, &codec->delayed_work,
msecs_to_jiffies(1000));
@@ -677,15 +640,14 @@ static int wm8971_resume(struct platform_device *pdev)
return 0;
}
-static int wm8971_init(struct snd_soc_device *socdev)
+static int wm8971_init(struct snd_soc_device *socdev,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = socdev->card->codec;
int reg, ret = 0;
codec->name = "WM8971";
codec->owner = THIS_MODULE;
- codec->read = wm8971_read_reg_cache;
- codec->write = wm8971_write;
codec->set_bias_level = wm8971_set_bias_level;
codec->dai = &wm8971_dai;
codec->reg_cache_size = ARRAY_SIZE(wm8971_reg);
@@ -695,42 +657,48 @@ static int wm8971_init(struct snd_soc_device *socdev)
if (codec->reg_cache == NULL)
return -ENOMEM;
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
wm8971_reset(codec);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "wm8971: failed to create pcms\n");
- goto pcm_err;
+ goto err;
}
/* charge output caps - set vmid to 5k for quick power up */
- reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
- wm8971_write(codec, WM8971_PWR1, reg | 0x01c0);
+ reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
+ snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
codec->bias_level = SND_SOC_BIAS_STANDBY;
queue_delayed_work(wm8971_workq, &codec->delayed_work,
msecs_to_jiffies(1000));
/* set the update bits */
- reg = wm8971_read_reg_cache(codec, WM8971_LDAC);
- wm8971_write(codec, WM8971_LDAC, reg | 0x0100);
- reg = wm8971_read_reg_cache(codec, WM8971_RDAC);
- wm8971_write(codec, WM8971_RDAC, reg | 0x0100);
-
- reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V);
- wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100);
- reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V);
- wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100);
-
- reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V);
- wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100);
- reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V);
- wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100);
-
- reg = wm8971_read_reg_cache(codec, WM8971_LINVOL);
- wm8971_write(codec, WM8971_LINVOL, reg | 0x0100);
- reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
- wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8971_LDAC);
+ snd_soc_write(codec, WM8971_LDAC, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8971_RDAC);
+ snd_soc_write(codec, WM8971_RDAC, reg | 0x0100);
+
+ reg = snd_soc_read(codec, WM8971_LOUT1V);
+ snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8971_ROUT1V);
+ snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100);
+
+ reg = snd_soc_read(codec, WM8971_LOUT2V);
+ snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8971_ROUT2V);
+ snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100);
+
+ reg = snd_soc_read(codec, WM8971_LINVOL);
+ snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8971_RINVOL);
+ snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100);
snd_soc_add_controls(codec, wm8971_snd_controls,
ARRAY_SIZE(wm8971_snd_controls));
@@ -745,7 +713,7 @@ static int wm8971_init(struct snd_soc_device *socdev)
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-pcm_err:
+err:
kfree(codec->reg_cache);
return ret;
}
@@ -767,7 +735,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c,
codec->control_data = i2c;
- ret = wm8971_init(socdev);
+ ret = wm8971_init(socdev, SND_SOC_I2C);
if (ret < 0)
pr_err("failed to initialise WM8971\n");
@@ -877,7 +845,6 @@ static int wm8971_probe(struct platform_device *pdev)
#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
if (setup->i2c_address) {
- codec->hw_write = (hw_write_t)i2c_master_send;
ret = wm8971_add_i2c_device(pdev, setup);
}
#endif
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
new file mode 100644
index 0000000..d8a013a
--- /dev/null
+++ b/sound/soc/codecs/wm8974.c
@@ -0,0 +1,808 @@
+/*
+ * wm8974.c -- WM8974 ALSA Soc Audio driver
+ *
+ * Copyright 2006-2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <linux@wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8974.h"
+
+static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0050, 0x0000, 0x0140, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x00ff,
+ 0x0000, 0x0000, 0x0100, 0x00ff,
+ 0x0000, 0x0000, 0x012c, 0x002c,
+ 0x002c, 0x002c, 0x002c, 0x0000,
+ 0x0032, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0038, 0x000b, 0x0032, 0x0000,
+ 0x0008, 0x000c, 0x0093, 0x00e9,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0003, 0x0010, 0x0000, 0x0000,
+ 0x0000, 0x0002, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0039, 0x0000,
+ 0x0000,
+};
+
+#define WM8974_POWER1_BIASEN 0x08
+#define WM8974_POWER1_BUFIOEN 0x10
+
+struct wm8974_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM8974_CACHEREGNUM];
+};
+
+static struct snd_soc_codec *wm8974_codec;
+
+#define wm8974_reset(c) snd_soc_write(c, WM8974_RESET, 0)
+
+static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" };
+static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8974_eqmode[] = {"Capture", "Playback" };
+static const char *wm8974_bw[] = {"Narrow", "Wide" };
+static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
+static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
+static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
+static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
+static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
+static const char *wm8974_alc[] = {"ALC", "Limiter" };
+
+static const struct soc_enum wm8974_enum[] = {
+ SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */
+ SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */
+ SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp),
+ SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode),
+
+ SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1),
+ SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw),
+ SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2),
+ SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw),
+
+ SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3),
+ SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw),
+ SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4),
+ SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw),
+
+ SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5),
+ SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc),
+};
+
+static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" };
+
+static const struct soc_enum wm8974_auxmode =
+ SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
+
+static const struct snd_kcontrol_new wm8974_snd_controls[] = {
+
+SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0),
+
+SOC_ENUM("DAC Companding", wm8974_enum[1]),
+SOC_ENUM("ADC Companding", wm8974_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", wm8974_enum[2]),
+SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0),
+
+SOC_SINGLE_TLV("PCM Volume", WM8974_DACVOL, 0, 255, 0, digital_tlv),
+
+SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0),
+SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0),
+SOC_SINGLE("ADC Inversion Switch", WM8974_ADC, 0, 1, 0),
+
+SOC_SINGLE_TLV("Capture Volume", WM8974_ADCVOL, 0, 255, 0, digital_tlv),
+
+SOC_ENUM("Equaliser Function", wm8974_enum[3]),
+SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]),
+SOC_SINGLE_TLV("EQ1 Volume", WM8974_EQ1, 0, 24, 1, eq_tlv),
+
+SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]),
+SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]),
+SOC_SINGLE_TLV("EQ2 Volume", WM8974_EQ2, 0, 24, 1, eq_tlv),
+
+SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]),
+SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]),
+SOC_SINGLE_TLV("EQ3 Volume", WM8974_EQ3, 0, 24, 1, eq_tlv),
+
+SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]),
+SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]),
+SOC_SINGLE_TLV("EQ4 Volume", WM8974_EQ4, 0, 24, 1, eq_tlv),
+
+SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]),
+SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]),
+SOC_SINGLE_TLV("EQ5 Volume", WM8974_EQ5, 0, 24, 1, eq_tlv),
+
+SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0),
+SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0),
+SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0),
+
+SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0),
+SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0),
+
+SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0),
+SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0),
+SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0),
+
+SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0),
+SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0),
+
+SOC_ENUM("ALC Capture Mode", wm8974_enum[13]),
+SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0),
+
+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0),
+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0),
+
+SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0),
+SOC_SINGLE_TLV("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0, inpga_tlv),
+
+SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1),
+SOC_SINGLE_TLV("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0, spk_tlv),
+
+SOC_ENUM("Aux Mode", wm8974_auxmode),
+
+SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0),
+SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1),
+};
+
+/* Speaker Output Mixer */
+static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1),
+};
+
+/* Mono Output Mixer */
+static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
+};
+
+/* Boost mixer */
+static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+};
+
+/* Input PGA */
+static const struct snd_kcontrol_new wm8974_inpga[] = {
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPUT, 2, 1, 0),
+SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0),
+SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0),
+};
+
+/* AUX Input boost vol */
+static const struct snd_kcontrol_new wm8974_aux_boost_controls =
+SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
+
+/* Mic Input boost vol */
+static const struct snd_kcontrol_new wm8974_mic_boost_controls =
+SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
+
+static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
+ &wm8974_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm8974_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0,
+ &wm8974_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8974_mono_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER2, 0, 0),
+SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0),
+
+SND_SOC_DAPM_MIXER("Input PGA", WM8974_POWER2, 2, 0, wm8974_inpga,
+ ARRAY_SIZE(wm8974_inpga)),
+SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0,
+ wm8974_boost_mixer, ARRAY_SIZE(wm8974_boost_mixer)),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0),
+
+SND_SOC_DAPM_INPUT("MICN"),
+SND_SOC_DAPM_INPUT("MICP"),
+SND_SOC_DAPM_INPUT("AUX"),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Mono output mixer */
+ {"Mono Mixer", "PCM Playback Switch", "DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Speaker output mixer */
+ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Outputs */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono Out"},
+ {"SpkN Out", NULL, "Speaker Mixer"},
+ {"SpkP Out", NULL, "Speaker Mixer"},
+ {"SPKOUTN", NULL, "SpkN Out"},
+ {"SPKOUTP", NULL, "SpkP Out"},
+
+ /* Boost Mixer */
+ {"ADC", NULL, "Boost Mixer"},
+ {"Boost Mixer", "Aux Switch", "Aux Input"},
+ {"Boost Mixer", NULL, "Input PGA"},
+ {"Boost Mixer", NULL, "MICP"},
+
+ /* Input PGA */
+ {"Input PGA", "Aux Switch", "Aux Input"},
+ {"Input PGA", "MicN Switch", "MICN"},
+ {"Input PGA", "MicP Switch", "MICP"},
+
+ /* Inputs */
+ {"Aux Input", NULL, "AUX"},
+};
+
+static int wm8974_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8974_dapm_widgets,
+ ARRAY_SIZE(wm8974_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct pll_ {
+ unsigned int pre_div:4; /* prescale - 1 */
+ unsigned int n:4;
+ unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static void pll_factors(unsigned int target, unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div.pre_div = 1;
+ Ndiv = target / source;
+ } else
+ pll_div.pre_div = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8974 N value %u outwith recommended range!\n",
+ Ndiv);
+
+ pll_div.n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div.k = K;
+}
+
+static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ if (freq_in == 0 || freq_out == 0) {
+ /* Clock CODEC directly from MCLK */
+ reg = snd_soc_read(codec, WM8974_CLOCK);
+ snd_soc_write(codec, WM8974_CLOCK, reg & 0x0ff);
+
+ /* Turn off PLL */
+ reg = snd_soc_read(codec, WM8974_POWER1);
+ snd_soc_write(codec, WM8974_POWER1, reg & 0x1df);
+ return 0;
+ }
+
+ pll_factors(freq_out*4, freq_in);
+
+ snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n);
+ snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18);
+ snd_soc_write(codec, WM8974_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ snd_soc_write(codec, WM8974_PLLK3, pll_div.k & 0x1ff);
+ reg = snd_soc_read(codec, WM8974_POWER1);
+ snd_soc_write(codec, WM8974_POWER1, reg | 0x020);
+
+ /* Run CODEC from PLL instead of MCLK */
+ reg = snd_soc_read(codec, WM8974_CLOCK);
+ snd_soc_write(codec, WM8974_CLOCK, reg | 0x100);
+
+ return 0;
+}
+
+/*
+ * Configure WM8974 clock dividers.
+ */
+static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8974_OPCLKDIV:
+ reg = snd_soc_read(codec, WM8974_GPIO) & 0x1cf;
+ snd_soc_write(codec, WM8974_GPIO, reg | div);
+ break;
+ case WM8974_MCLKDIV:
+ reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f;
+ snd_soc_write(codec, WM8974_CLOCK, reg | div);
+ break;
+ case WM8974_ADCCLK:
+ reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7;
+ snd_soc_write(codec, WM8974_ADC, reg | div);
+ break;
+ case WM8974_DACCLK:
+ reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7;
+ snd_soc_write(codec, WM8974_DAC, reg | div);
+ break;
+ case WM8974_BCLKDIV:
+ reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3;
+ snd_soc_write(codec, WM8974_CLOCK, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+ u16 clk = snd_soc_read(codec, WM8974_CLOCK) & 0x1fe;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clk |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0010;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0008;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x00018;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0180;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0100;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0080;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, WM8974_IFACE, iface);
+ snd_soc_write(codec, WM8974_CLOCK, clk);
+ return 0;
+}
+
+static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 iface = snd_soc_read(codec, WM8974_IFACE) & 0x19f;
+ u16 adn = snd_soc_read(codec, WM8974_ADD) & 0x1f1;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0020;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0040;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x0060;
+ break;
+ }
+
+ /* filter coefficient */
+ switch (params_rate(params)) {
+ case SNDRV_PCM_RATE_8000:
+ adn |= 0x5 << 1;
+ break;
+ case SNDRV_PCM_RATE_11025:
+ adn |= 0x4 << 1;
+ break;
+ case SNDRV_PCM_RATE_16000:
+ adn |= 0x3 << 1;
+ break;
+ case SNDRV_PCM_RATE_22050:
+ adn |= 0x2 << 1;
+ break;
+ case SNDRV_PCM_RATE_32000:
+ adn |= 0x1 << 1;
+ break;
+ case SNDRV_PCM_RATE_44100:
+ case SNDRV_PCM_RATE_48000:
+ break;
+ }
+
+ snd_soc_write(codec, WM8974_IFACE, iface);
+ snd_soc_write(codec, WM8974_ADD, adn);
+ return 0;
+}
+
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+ else
+ snd_soc_write(codec, WM8974_DAC, mute_reg);
+ return 0;
+}
+
+/* liam need to make this lower power with dapm */
+static int wm8974_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 power1 = snd_soc_read(codec, WM8974_POWER1) & ~0x3;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ power1 |= 0x1; /* VMID 50k */
+ snd_soc_write(codec, WM8974_POWER1, power1);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
+
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Initial cap charge at VMID 5k */
+ snd_soc_write(codec, WM8974_POWER1, power1 | 0x3);
+ mdelay(100);
+ }
+
+ power1 |= 0x2; /* VMID 500k */
+ snd_soc_write(codec, WM8974_POWER1, power1);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_write(codec, WM8974_POWER1, 0);
+ snd_soc_write(codec, WM8974_POWER2, 0);
+ snd_soc_write(codec, WM8974_POWER3, 0);
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8974_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8974_ops = {
+ .hw_params = wm8974_pcm_hw_params,
+ .digital_mute = wm8974_mute,
+ .set_fmt = wm8974_set_dai_fmt,
+ .set_clkdiv = wm8974_set_dai_clkdiv,
+ .set_pll = wm8974_set_dai_pll,
+};
+
+struct snd_soc_dai wm8974_dai = {
+ .name = "WM8974 HiFi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2, /* Only 1 channel of data */
+ .rates = WM8974_RATES,
+ .formats = WM8974_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2, /* Only 1 channel of data */
+ .rates = WM8974_RATES,
+ .formats = WM8974_FORMATS,},
+ .ops = &wm8974_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8974_dai);
+
+static int wm8974_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8974_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8974_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+static int wm8974_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8974_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8974_codec;
+ codec = wm8974_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8974_snd_controls,
+ ARRAY_SIZE(wm8974_snd_controls));
+ wm8974_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8974_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8974 = {
+ .probe = wm8974_probe,
+ .remove = wm8974_remove,
+ .suspend = wm8974_suspend,
+ .resume = wm8974_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974);
+
+static __devinit int wm8974_register(struct wm8974_priv *wm8974)
+{
+ int ret;
+ struct snd_soc_codec *codec = &wm8974->codec;
+
+ if (wm8974_codec) {
+ dev_err(codec->dev, "Another WM8974 is registered\n");
+ return -EINVAL;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8974;
+ codec->name = "WM8974";
+ codec->owner = THIS_MODULE;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8974_set_bias_level;
+ codec->dai = &wm8974_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8974_CACHEREGNUM;
+ codec->reg_cache = &wm8974->reg_cache;
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
+ memcpy(codec->reg_cache, wm8974_reg, sizeof(wm8974_reg));
+
+ ret = wm8974_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ goto err;
+ }
+
+ wm8974_dai.dev = codec->dev;
+
+ wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ wm8974_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ ret = snd_soc_register_dai(&wm8974_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ goto err_codec;
+ }
+
+ return 0;
+
+err_codec:
+ snd_soc_unregister_codec(codec);
+err:
+ kfree(wm8974);
+ return ret;
+}
+
+static __devexit void wm8974_unregister(struct wm8974_priv *wm8974)
+{
+ wm8974_set_bias_level(&wm8974->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8974_dai);
+ snd_soc_unregister_codec(&wm8974->codec);
+ kfree(wm8974);
+ wm8974_codec = NULL;
+}
+
+static __devinit int wm8974_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8974_priv *wm8974;
+ struct snd_soc_codec *codec;
+
+ wm8974 = kzalloc(sizeof(struct wm8974_priv), GFP_KERNEL);
+ if (wm8974 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8974->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8974);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8974_register(wm8974);
+}
+
+static __devexit int wm8974_i2c_remove(struct i2c_client *client)
+{
+ struct wm8974_priv *wm8974 = i2c_get_clientdata(client);
+ wm8974_unregister(wm8974);
+ return 0;
+}
+
+static const struct i2c_device_id wm8974_i2c_id[] = {
+ { "wm8974", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id);
+
+static struct i2c_driver wm8974_i2c_driver = {
+ .driver = {
+ .name = "WM8974",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8974_i2c_probe,
+ .remove = __devexit_p(wm8974_i2c_remove),
+ .id_table = wm8974_i2c_id,
+};
+
+static int __init wm8974_modinit(void)
+{
+ return i2c_add_driver(&wm8974_i2c_driver);
+}
+module_init(wm8974_modinit);
+
+static void __exit wm8974_exit(void)
+{
+ i2c_del_driver(&wm8974_i2c_driver);
+}
+module_exit(wm8974_exit);
+
+MODULE_DESCRIPTION("ASoC WM8974 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h
new file mode 100644
index 0000000..98de956
--- /dev/null
+++ b/sound/soc/codecs/wm8974.h
@@ -0,0 +1,99 @@
+/*
+ * wm8974.h -- WM8974 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8974_H
+#define _WM8974_H
+
+/* WM8974 register space */
+
+#define WM8974_RESET 0x0
+#define WM8974_POWER1 0x1
+#define WM8974_POWER2 0x2
+#define WM8974_POWER3 0x3
+#define WM8974_IFACE 0x4
+#define WM8974_COMP 0x5
+#define WM8974_CLOCK 0x6
+#define WM8974_ADD 0x7
+#define WM8974_GPIO 0x8
+#define WM8974_DAC 0xa
+#define WM8974_DACVOL 0xb
+#define WM8974_ADC 0xe
+#define WM8974_ADCVOL 0xf
+#define WM8974_EQ1 0x12
+#define WM8974_EQ2 0x13
+#define WM8974_EQ3 0x14
+#define WM8974_EQ4 0x15
+#define WM8974_EQ5 0x16
+#define WM8974_DACLIM1 0x18
+#define WM8974_DACLIM2 0x19
+#define WM8974_NOTCH1 0x1b
+#define WM8974_NOTCH2 0x1c
+#define WM8974_NOTCH3 0x1d
+#define WM8974_NOTCH4 0x1e
+#define WM8974_ALC1 0x20
+#define WM8974_ALC2 0x21
+#define WM8974_ALC3 0x22
+#define WM8974_NGATE 0x23
+#define WM8974_PLLN 0x24
+#define WM8974_PLLK1 0x25
+#define WM8974_PLLK2 0x26
+#define WM8974_PLLK3 0x27
+#define WM8974_ATTEN 0x28
+#define WM8974_INPUT 0x2c
+#define WM8974_INPPGA 0x2d
+#define WM8974_ADCBOOST 0x2f
+#define WM8974_OUTPUT 0x31
+#define WM8974_SPKMIX 0x32
+#define WM8974_SPKVOL 0x36
+#define WM8974_MONOMIX 0x38
+
+#define WM8974_CACHEREGNUM 57
+
+/* Clock divider Id's */
+#define WM8974_OPCLKDIV 0
+#define WM8974_MCLKDIV 1
+#define WM8974_ADCCLK 2
+#define WM8974_DACCLK 3
+#define WM8974_BCLKDIV 4
+
+/* DAC clock dividers */
+#define WM8974_DACCLK_F2 (1 << 3)
+#define WM8974_DACCLK_F4 (0 << 3)
+
+/* ADC clock dividers */
+#define WM8974_ADCCLK_F2 (1 << 3)
+#define WM8974_ADCCLK_F4 (0 << 3)
+
+/* PLL Out dividers */
+#define WM8974_OPCLKDIV_1 (0 << 4)
+#define WM8974_OPCLKDIV_2 (1 << 4)
+#define WM8974_OPCLKDIV_3 (2 << 4)
+#define WM8974_OPCLKDIV_4 (3 << 4)
+
+/* BCLK clock dividers */
+#define WM8974_BCLKDIV_1 (0 << 2)
+#define WM8974_BCLKDIV_2 (1 << 2)
+#define WM8974_BCLKDIV_4 (2 << 2)
+#define WM8974_BCLKDIV_8 (3 << 2)
+#define WM8974_BCLKDIV_16 (4 << 2)
+#define WM8974_BCLKDIV_32 (5 << 2)
+
+/* MCLK clock dividers */
+#define WM8974_MCLKDIV_1 (0 << 5)
+#define WM8974_MCLKDIV_1_5 (1 << 5)
+#define WM8974_MCLKDIV_2 (2 << 5)
+#define WM8974_MCLKDIV_3 (3 << 5)
+#define WM8974_MCLKDIV_4 (4 << 5)
+#define WM8974_MCLKDIV_6 (5 << 5)
+#define WM8974_MCLKDIV_8 (6 << 5)
+#define WM8974_MCLKDIV_12 (7 << 5)
+
+extern struct snd_soc_dai wm8974_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8974;
+
+#endif
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 8c0fdf8..3f530f8 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -57,50 +57,7 @@ struct wm8988_priv {
};
-/*
- * read wm8988 register cache
- */
-static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- if (reg > WM8988_NUM_REG)
- return -1;
- return cache[reg];
-}
-
-/*
- * write wm8988 register cache
- */
-static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- if (reg > WM8988_NUM_REG)
- return;
- cache[reg] = value;
-}
-
-static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[2];
-
- /* data is
- * D15..D9 WM8753 register offset
- * D8...D0 register data
- */
- data[0] = (reg << 1) | ((value >> 8) & 0x0001);
- data[1] = value & 0x00ff;
-
- wm8988_write_reg_cache(codec, reg, value);
- if (codec->hw_write(codec->control_data, data, 2) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0)
+#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0)
/*
* WM8988 Controls
@@ -226,15 +183,15 @@ static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2);
+ u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2);
/* Use the DAC to gate LRC if active, otherwise use ADC */
- if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180)
+ if (snd_soc_read(codec, WM8988_PWR2) & 0x180)
adctl2 &= ~0x4;
else
adctl2 |= 0x4;
- return wm8988_write(codec, WM8988_ADCTL2, adctl2);
+ return snd_soc_write(codec, WM8988_ADCTL2, adctl2);
}
static const char *wm8988_line_texts[] = {
@@ -619,7 +576,7 @@ static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8988_write(codec, WM8988_IFACE, iface);
+ snd_soc_write(codec, WM8988_IFACE, iface);
return 0;
}
@@ -653,8 +610,8 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct wm8988_priv *wm8988 = codec->private_data;
- u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3;
- u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180;
+ u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3;
+ u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180;
int coeff;
coeff = get_coeff(wm8988->sysclk, params_rate(params));
@@ -685,9 +642,9 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
}
/* set iface & srate */
- wm8988_write(codec, WM8988_IFACE, iface);
+ snd_soc_write(codec, WM8988_IFACE, iface);
if (coeff >= 0)
- wm8988_write(codec, WM8988_SRATE, srate |
+ snd_soc_write(codec, WM8988_SRATE, srate |
(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
return 0;
@@ -696,19 +653,19 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
static int wm8988_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7;
+ u16 mute_reg = snd_soc_read(codec, WM8988_ADCDAC) & 0xfff7;
if (mute)
- wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
+ snd_soc_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
else
- wm8988_write(codec, WM8988_ADCDAC, mute_reg);
+ snd_soc_write(codec, WM8988_ADCDAC, mute_reg);
return 0;
}
static int wm8988_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1;
+ u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1;
switch (level) {
case SND_SOC_BIAS_ON:
@@ -716,24 +673,24 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
/* VREF, VMID=2x50k, digital enabled */
- wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
+ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* VREF, VMID=2x5k */
- wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
+ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
/* Charge caps */
msleep(100);
}
/* VREF, VMID=2*500k, digital stopped */
- wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
+ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
break;
case SND_SOC_BIAS_OFF:
- wm8988_write(codec, WM8988_PWR1, 0x0000);
+ snd_soc_write(codec, WM8988_PWR1, 0x0000);
break;
}
codec->bias_level = level;
@@ -868,7 +825,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8988 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988);
-static int wm8988_register(struct wm8988_priv *wm8988)
+static int wm8988_register(struct wm8988_priv *wm8988,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = &wm8988->codec;
int ret;
@@ -887,8 +845,6 @@ static int wm8988_register(struct wm8988_priv *wm8988)
codec->private_data = wm8988;
codec->name = "WM8988";
codec->owner = THIS_MODULE;
- codec->read = wm8988_read_reg_cache;
- codec->write = wm8988_write;
codec->dai = &wm8988_dai;
codec->num_dai = 1;
codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache);
@@ -899,23 +855,29 @@ static int wm8988_register(struct wm8988_priv *wm8988)
memcpy(codec->reg_cache, wm8988_reg,
sizeof(wm8988_reg));
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err;
+ }
+
ret = wm8988_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
- return ret;
+ goto err;
}
/* set the update bits (we always update left then right) */
- reg = wm8988_read_reg_cache(codec, WM8988_RADC);
- wm8988_write(codec, WM8988_RADC, reg | 0x100);
- reg = wm8988_read_reg_cache(codec, WM8988_RDAC);
- wm8988_write(codec, WM8988_RDAC, reg | 0x0100);
- reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V);
- wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100);
- reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V);
- wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100);
- reg = wm8988_read_reg_cache(codec, WM8988_RINVOL);
- wm8988_write(codec, WM8988_RINVOL, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8988_RADC);
+ snd_soc_write(codec, WM8988_RADC, reg | 0x100);
+ reg = snd_soc_read(codec, WM8988_RDAC);
+ snd_soc_write(codec, WM8988_RDAC, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8988_ROUT1V);
+ snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8988_ROUT2V);
+ snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100);
+ reg = snd_soc_read(codec, WM8988_RINVOL);
+ snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100);
wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY);
@@ -926,18 +888,20 @@ static int wm8988_register(struct wm8988_priv *wm8988)
ret = snd_soc_register_codec(codec);
if (ret != 0) {
dev_err(codec->dev, "Failed to register codec: %d\n", ret);
- return ret;
+ goto err;
}
ret = snd_soc_register_dai(&wm8988_dai);
if (ret != 0) {
dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
snd_soc_unregister_codec(codec);
- return ret;
+ goto err_codec;
}
return 0;
+err_codec:
+ snd_soc_unregister_codec(codec);
err:
kfree(wm8988);
return ret;
@@ -964,14 +928,13 @@ static int wm8988_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
codec = &wm8988->codec;
- codec->hw_write = (hw_write_t)i2c_master_send;
i2c_set_clientdata(i2c, wm8988);
codec->control_data = i2c;
codec->dev = &i2c->dev;
- return wm8988_register(wm8988);
+ return wm8988_register(wm8988, SND_SOC_I2C);
}
static int wm8988_i2c_remove(struct i2c_client *client)
@@ -981,6 +944,21 @@ static int wm8988_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8988_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8988_i2c_suspend NULL
+#define wm8988_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm8988_i2c_id[] = {
{ "wm8988", 0 },
{ }
@@ -994,35 +972,13 @@ static struct i2c_driver wm8988_i2c_driver = {
},
.probe = wm8988_i2c_probe,
.remove = wm8988_i2c_remove,
+ .suspend = wm8988_i2c_suspend,
+ .resume = wm8988_i2c_resume,
.id_table = wm8988_i2c_id,
};
#endif
#if defined(CONFIG_SPI_MASTER)
-static int wm8988_spi_write(struct spi_device *spi, const char *data, int len)
-{
- struct spi_transfer t;
- struct spi_message m;
- u8 msg[2];
-
- if (len <= 0)
- return 0;
-
- msg[0] = data[0];
- msg[1] = data[1];
-
- spi_message_init(&m);
- memset(&t, 0, (sizeof t));
-
- t.tx_buf = &msg[0];
- t.len = len;
-
- spi_message_add_tail(&t, &m);
- spi_sync(spi, &m);
-
- return len;
-}
-
static int __devinit wm8988_spi_probe(struct spi_device *spi)
{
struct wm8988_priv *wm8988;
@@ -1033,13 +989,12 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi)
return -ENOMEM;
codec = &wm8988->codec;
- codec->hw_write = (hw_write_t)wm8988_spi_write;
codec->control_data = spi;
codec->dev = &spi->dev;
dev_set_drvdata(&spi->dev, wm8988);
- return wm8988_register(wm8988);
+ return wm8988_register(wm8988, SND_SOC_SPI);
}
static int __devexit wm8988_spi_remove(struct spi_device *spi)
@@ -1051,6 +1006,21 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi)
return 0;
}
+#ifdef CONFIG_PM
+static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8988_spi_resume(struct spi_device *spi)
+{
+ return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8988_spi_suspend NULL
+#define wm8988_spi_resume NULL
+#endif
+
static struct spi_driver wm8988_spi_driver = {
.driver = {
.name = "wm8988",
@@ -1059,6 +1029,8 @@ static struct spi_driver wm8988_spi_driver = {
},
.probe = wm8988_spi_probe,
.remove = __devexit_p(wm8988_spi_remove),
+ .suspend = wm8988_spi_suspend,
+ .resume = wm8988_spi_resume,
};
#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index d029818..2d702db 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -108,53 +108,7 @@ static const u16 wm8990_reg[] = {
0x0000, /* R63 - Driver internal */
};
-/*
- * read wm8990 register cache
- */
-static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- BUG_ON(reg >= ARRAY_SIZE(wm8990_reg));
- return cache[reg];
-}
-
-/*
- * write wm8990 register cache
- */
-static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int value)
-{
- u16 *cache = codec->reg_cache;
-
- /* Reset register and reserved registers are uncached */
- if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg))
- return;
-
- cache[reg] = value;
-}
-
-/*
- * write to the wm8990 register space
- */
-static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 data[3];
-
- data[0] = reg & 0xFF;
- data[1] = (value >> 8) & 0xFF;
- data[2] = value & 0xFF;
-
- wm8990_write_reg_cache(codec, reg, value);
-
- if (codec->hw_write(codec->control_data, data, 3) == 2)
- return 0;
- else
- return -EIO;
-}
-
-#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0)
+#define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0)
static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
@@ -187,8 +141,8 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
return ret;
/* now hit the volume update bits (always bit 8) */
- val = wm8990_read_reg_cache(codec, reg);
- return wm8990_write(codec, reg, val | 0x0100);
+ val = snd_soc_read(codec, reg);
+ return snd_soc_write(codec, reg, val | 0x0100);
}
#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\
@@ -427,8 +381,8 @@ static int inmixer_event(struct snd_soc_dapm_widget *w,
{
u16 reg, fakepower;
- reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2);
- fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS);
+ reg = snd_soc_read(w->codec, WM8990_POWER_MANAGEMENT_2);
+ fakepower = snd_soc_read(w->codec, WM8990_INTDRIVBITS);
if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) |
(1 << WM8990_AINLMUX_PWR_BIT))) {
@@ -443,7 +397,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w,
} else {
reg &= ~WM8990_AINL_ENA;
}
- wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
+ snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
return 0;
}
@@ -457,7 +411,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
switch (reg_shift) {
case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) :
- reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1);
+ reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER1);
if (reg & WM8990_LDLO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 1 LDLO Set\n");
@@ -465,7 +419,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8):
- reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2);
+ reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER2);
if (reg & WM8990_RDRO) {
printk(KERN_WARNING
"Cannot set as Output Mixer 2 RDRO Set\n");
@@ -473,7 +427,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8):
- reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER);
if (reg & WM8990_LDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer LDSPK Set\n");
@@ -481,7 +435,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
}
break;
case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8):
- reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER);
if (reg & WM8990_RDSPK) {
printk(KERN_WARNING
"Cannot set as Speaker Mixer RDSPK Set\n");
@@ -1029,24 +983,24 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
pll_factors(&pll_div, freq_out * 4, freq_in);
/* Turn on PLL */
- reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
reg |= WM8990_PLL_ENA;
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
/* sysclk comes from PLL */
- reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2);
- wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
+ reg = snd_soc_read(codec, WM8990_CLOCKING_2);
+ snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
/* set up N , fractional mode and pre-divisor if neccessary */
- wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
+ snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
(pll_div.div2?WM8990_PRESCALE:0));
- wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
- wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
+ snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
+ snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
} else {
/* Turn on PLL */
- reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
reg &= ~WM8990_PLL_ENA;
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
}
return 0;
}
@@ -1073,8 +1027,8 @@ static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
struct snd_soc_codec *codec = codec_dai->codec;
u16 audio1, audio3;
- audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
- audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3);
+ audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1);
+ audio3 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_3);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1115,8 +1069,8 @@ static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
- wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3);
+ snd_soc_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ snd_soc_write(codec, WM8990_AUDIO_INTERFACE_3, audio3);
return 0;
}
@@ -1128,24 +1082,24 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
switch (div_id) {
case WM8990_MCLK_DIV:
- reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
~WM8990_MCLK_DIV_MASK;
- wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
break;
case WM8990_DACCLK_DIV:
- reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
~WM8990_DAC_CLKDIV_MASK;
- wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
break;
case WM8990_ADCCLK_DIV:
- reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
~WM8990_ADC_CLKDIV_MASK;
- wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
break;
case WM8990_BCLK_DIV:
- reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) &
+ reg = snd_soc_read(codec, WM8990_CLOCKING_1) &
~WM8990_BCLK_DIV_MASK;
- wm8990_write(codec, WM8990_CLOCKING_1, reg | div);
+ snd_soc_write(codec, WM8990_CLOCKING_1, reg | div);
break;
default:
return -EINVAL;
@@ -1164,7 +1118,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
- u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+ u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1);
audio1 &= ~WM8990_AIF_WL_MASK;
/* bit size */
@@ -1182,7 +1136,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
break;
}
- wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ snd_soc_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
return 0;
}
@@ -1191,12 +1145,12 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute)
struct snd_soc_codec *codec = dai->codec;
u16 val;
- val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
+ val = snd_soc_read(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
if (mute)
- wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+ snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
else
- wm8990_write(codec, WM8990_DAC_CTRL, val);
+ snd_soc_write(codec, WM8990_DAC_CTRL, val);
return 0;
}
@@ -1212,21 +1166,21 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
/* VMID=2*50k */
- val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) &
~WM8990_VMID_MODE_MASK;
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Enable all output discharge bits */
- wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
WM8990_DIS_ROUT);
/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
WM8990_BUFDCOPEN | WM8990_POBCTRL |
WM8990_VMIDTOG);
@@ -1234,83 +1188,83 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
msleep(msecs_to_jiffies(300));
/* Disable VMIDTOG */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
WM8990_BUFDCOPEN | WM8990_POBCTRL);
/* disable all output discharge bits */
- wm8990_write(codec, WM8990_ANTIPOP1, 0);
+ snd_soc_write(codec, WM8990_ANTIPOP1, 0);
/* Enable outputs */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
msleep(msecs_to_jiffies(50));
/* Enable VMID at 2x50k */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
msleep(msecs_to_jiffies(100));
/* Enable VREF */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
msleep(msecs_to_jiffies(600));
/* Enable BUFIOEN */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
WM8990_BUFDCOPEN | WM8990_POBCTRL |
WM8990_BUFIOEN);
/* Disable outputs */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3);
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
/* Enable workaround for ADC clocking issue. */
- wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
- wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
- wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
+ snd_soc_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+ snd_soc_write(codec, WM8990_EXT_CTL1, 0xa003);
+ snd_soc_write(codec, WM8990_EXT_ACCESS_ENA, 0);
}
/* VMID=2*250k */
- val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+ val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) &
~WM8990_VMID_MODE_MASK;
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
break;
case SND_SOC_BIAS_OFF:
/* Enable POBCTRL and SOFT_ST */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
WM8990_POBCTRL | WM8990_BUFIOEN);
/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
WM8990_BUFDCOPEN | WM8990_POBCTRL |
WM8990_BUFIOEN);
/* mute DAC */
- val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL);
- wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+ val = snd_soc_read(codec, WM8990_DAC_CTRL);
+ snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
/* Enable any disabled outputs */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
/* Disable VMID */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
msleep(msecs_to_jiffies(300));
/* Enable all output discharge bits */
- wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
WM8990_DIS_ROUT);
/* Disable VREF */
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0);
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
- wm8990_write(codec, WM8990_ANTIPOP2, 0x0);
+ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
break;
}
@@ -1411,8 +1365,6 @@ static int wm8990_init(struct snd_soc_device *socdev)
codec->name = "WM8990";
codec->owner = THIS_MODULE;
- codec->read = wm8990_read_reg_cache;
- codec->write = wm8990_write;
codec->set_bias_level = wm8990_set_bias_level;
codec->dai = &wm8990_dai;
codec->num_dai = 2;
@@ -1422,6 +1374,12 @@ static int wm8990_init(struct snd_soc_device *socdev)
if (codec->reg_cache == NULL)
return -ENOMEM;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret);
+ goto pcm_err;
+ }
+
wm8990_reset(codec);
/* register pcms */
@@ -1435,18 +1393,18 @@ static int wm8990_init(struct snd_soc_device *socdev)
codec->bias_level = SND_SOC_BIAS_OFF;
wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4);
- wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
+ reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4);
+ snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
- reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) &
+ reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) &
~WM8990_GPIO1_SEL_MASK;
- wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
+ snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
- reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
- wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
+ reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
+ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
- wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
- wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
snd_soc_add_controls(codec, wm8990_snd_controls,
ARRAY_SIZE(wm8990_snd_controls));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
new file mode 100644
index 0000000..d998799
--- /dev/null
+++ b/sound/soc/codecs/wm8993.c
@@ -0,0 +1,1675 @@
+/*
+ * wm8993.c -- WM8993 ALSA SoC audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/wm8993.h>
+
+#include "wm8993.h"
+#include "wm_hubs.h"
+
+static u16 wm8993_reg_defaults[WM8993_REGISTER_COUNT] = {
+ 0x8993, /* R0 - Software Reset */
+ 0x0000, /* R1 - Power Management (1) */
+ 0x6000, /* R2 - Power Management (2) */
+ 0x0000, /* R3 - Power Management (3) */
+ 0x4050, /* R4 - Audio Interface (1) */
+ 0x4000, /* R5 - Audio Interface (2) */
+ 0x01C8, /* R6 - Clocking 1 */
+ 0x0000, /* R7 - Clocking 2 */
+ 0x0000, /* R8 - Audio Interface (3) */
+ 0x0040, /* R9 - Audio Interface (4) */
+ 0x0004, /* R10 - DAC CTRL */
+ 0x00C0, /* R11 - Left DAC Digital Volume */
+ 0x00C0, /* R12 - Right DAC Digital Volume */
+ 0x0000, /* R13 - Digital Side Tone */
+ 0x0300, /* R14 - ADC CTRL */
+ 0x00C0, /* R15 - Left ADC Digital Volume */
+ 0x00C0, /* R16 - Right ADC Digital Volume */
+ 0x0000, /* R17 */
+ 0x0000, /* R18 - GPIO CTRL 1 */
+ 0x0010, /* R19 - GPIO1 */
+ 0x0000, /* R20 - IRQ_DEBOUNCE */
+ 0x0000, /* R21 */
+ 0x8000, /* R22 - GPIOCTRL 2 */
+ 0x0800, /* R23 - GPIO_POL */
+ 0x008B, /* R24 - Left Line Input 1&2 Volume */
+ 0x008B, /* R25 - Left Line Input 3&4 Volume */
+ 0x008B, /* R26 - Right Line Input 1&2 Volume */
+ 0x008B, /* R27 - Right Line Input 3&4 Volume */
+ 0x006D, /* R28 - Left Output Volume */
+ 0x006D, /* R29 - Right Output Volume */
+ 0x0066, /* R30 - Line Outputs Volume */
+ 0x0020, /* R31 - HPOUT2 Volume */
+ 0x0079, /* R32 - Left OPGA Volume */
+ 0x0079, /* R33 - Right OPGA Volume */
+ 0x0003, /* R34 - SPKMIXL Attenuation */
+ 0x0003, /* R35 - SPKMIXR Attenuation */
+ 0x0011, /* R36 - SPKOUT Mixers */
+ 0x0100, /* R37 - SPKOUT Boost */
+ 0x0079, /* R38 - Speaker Volume Left */
+ 0x0079, /* R39 - Speaker Volume Right */
+ 0x0000, /* R40 - Input Mixer2 */
+ 0x0000, /* R41 - Input Mixer3 */
+ 0x0000, /* R42 - Input Mixer4 */
+ 0x0000, /* R43 - Input Mixer5 */
+ 0x0000, /* R44 - Input Mixer6 */
+ 0x0000, /* R45 - Output Mixer1 */
+ 0x0000, /* R46 - Output Mixer2 */
+ 0x0000, /* R47 - Output Mixer3 */
+ 0x0000, /* R48 - Output Mixer4 */
+ 0x0000, /* R49 - Output Mixer5 */
+ 0x0000, /* R50 - Output Mixer6 */
+ 0x0000, /* R51 - HPOUT2 Mixer */
+ 0x0000, /* R52 - Line Mixer1 */
+ 0x0000, /* R53 - Line Mixer2 */
+ 0x0000, /* R54 - Speaker Mixer */
+ 0x0000, /* R55 - Additional Control */
+ 0x0000, /* R56 - AntiPOP1 */
+ 0x0000, /* R57 - AntiPOP2 */
+ 0x0000, /* R58 - MICBIAS */
+ 0x0000, /* R59 */
+ 0x0000, /* R60 - FLL Control 1 */
+ 0x0000, /* R61 - FLL Control 2 */
+ 0x0000, /* R62 - FLL Control 3 */
+ 0x2EE0, /* R63 - FLL Control 4 */
+ 0x0002, /* R64 - FLL Control 5 */
+ 0x2287, /* R65 - Clocking 3 */
+ 0x025F, /* R66 - Clocking 4 */
+ 0x0000, /* R67 - MW Slave Control */
+ 0x0000, /* R68 */
+ 0x0002, /* R69 - Bus Control 1 */
+ 0x0000, /* R70 - Write Sequencer 0 */
+ 0x0000, /* R71 - Write Sequencer 1 */
+ 0x0000, /* R72 - Write Sequencer 2 */
+ 0x0000, /* R73 - Write Sequencer 3 */
+ 0x0000, /* R74 - Write Sequencer 4 */
+ 0x0000, /* R75 - Write Sequencer 5 */
+ 0x1F25, /* R76 - Charge Pump 1 */
+ 0x0000, /* R77 */
+ 0x0000, /* R78 */
+ 0x0000, /* R79 */
+ 0x0000, /* R80 */
+ 0x0000, /* R81 - Class W 0 */
+ 0x0000, /* R82 */
+ 0x0000, /* R83 */
+ 0x0000, /* R84 - DC Servo 0 */
+ 0x054A, /* R85 - DC Servo 1 */
+ 0x0000, /* R86 */
+ 0x0000, /* R87 - DC Servo 3 */
+ 0x0000, /* R88 - DC Servo Readback 0 */
+ 0x0000, /* R89 - DC Servo Readback 1 */
+ 0x0000, /* R90 - DC Servo Readback 2 */
+ 0x0000, /* R91 */
+ 0x0000, /* R92 */
+ 0x0000, /* R93 */
+ 0x0000, /* R94 */
+ 0x0000, /* R95 */
+ 0x0100, /* R96 - Analogue HP 0 */
+ 0x0000, /* R97 */
+ 0x0000, /* R98 - EQ1 */
+ 0x000C, /* R99 - EQ2 */
+ 0x000C, /* R100 - EQ3 */
+ 0x000C, /* R101 - EQ4 */
+ 0x000C, /* R102 - EQ5 */
+ 0x000C, /* R103 - EQ6 */
+ 0x0FCA, /* R104 - EQ7 */
+ 0x0400, /* R105 - EQ8 */
+ 0x00D8, /* R106 - EQ9 */
+ 0x1EB5, /* R107 - EQ10 */
+ 0xF145, /* R108 - EQ11 */
+ 0x0B75, /* R109 - EQ12 */
+ 0x01C5, /* R110 - EQ13 */
+ 0x1C58, /* R111 - EQ14 */
+ 0xF373, /* R112 - EQ15 */
+ 0x0A54, /* R113 - EQ16 */
+ 0x0558, /* R114 - EQ17 */
+ 0x168E, /* R115 - EQ18 */
+ 0xF829, /* R116 - EQ19 */
+ 0x07AD, /* R117 - EQ20 */
+ 0x1103, /* R118 - EQ21 */
+ 0x0564, /* R119 - EQ22 */
+ 0x0559, /* R120 - EQ23 */
+ 0x4000, /* R121 - EQ24 */
+ 0x0000, /* R122 - Digital Pulls */
+ 0x0F08, /* R123 - DRC Control 1 */
+ 0x0000, /* R124 - DRC Control 2 */
+ 0x0080, /* R125 - DRC Control 3 */
+ 0x0000, /* R126 - DRC Control 4 */
+};
+
+static struct {
+ int ratio;
+ int clk_sys_rate;
+} clk_sys_rates[] = {
+ { 64, 0 },
+ { 128, 1 },
+ { 192, 2 },
+ { 256, 3 },
+ { 384, 4 },
+ { 512, 5 },
+ { 768, 6 },
+ { 1024, 7 },
+ { 1408, 8 },
+ { 1536, 9 },
+};
+
+static struct {
+ int rate;
+ int sample_rate;
+} sample_rates[] = {
+ { 8000, 0 },
+ { 11025, 1 },
+ { 12000, 1 },
+ { 16000, 2 },
+ { 22050, 3 },
+ { 24000, 3 },
+ { 32000, 4 },
+ { 44100, 5 },
+ { 48000, 5 },
+};
+
+static struct {
+ int div; /* *10 due to .5s */
+ int bclk_div;
+} bclk_divs[] = {
+ { 10, 0 },
+ { 15, 1 },
+ { 20, 2 },
+ { 30, 3 },
+ { 40, 4 },
+ { 55, 5 },
+ { 60, 6 },
+ { 80, 7 },
+ { 110, 8 },
+ { 120, 9 },
+ { 160, 10 },
+ { 220, 11 },
+ { 240, 12 },
+ { 320, 13 },
+ { 440, 14 },
+ { 480, 15 },
+};
+
+struct wm8993_priv {
+ u16 reg_cache[WM8993_REGISTER_COUNT];
+ struct wm8993_platform_data pdata;
+ struct snd_soc_codec codec;
+ int master;
+ int sysclk_source;
+ int tdm_slots;
+ int tdm_width;
+ unsigned int mclk_rate;
+ unsigned int sysclk_rate;
+ unsigned int fs;
+ unsigned int bclk;
+ int class_w_users;
+ unsigned int fll_fref;
+ unsigned int fll_fout;
+};
+
+static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg)
+{
+ struct i2c_msg xfer[2];
+ u16 data;
+ int ret;
+ struct i2c_client *i2c = codec->control_data;
+
+ /* Write register */
+ xfer[0].addr = i2c->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = &reg;
+
+ /* Read data */
+ xfer[1].addr = i2c->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(i2c->adapter, xfer, 2);
+ if (ret != 2) {
+ dev_err(codec->dev, "Failed to read 0x%x: %d\n", reg, ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static int wm8993_volatile(unsigned int reg)
+{
+ switch (reg) {
+ case WM8993_SOFTWARE_RESET:
+ case WM8993_DC_SERVO_0:
+ case WM8993_DC_SERVO_READBACK_0:
+ case WM8993_DC_SERVO_READBACK_1:
+ case WM8993_DC_SERVO_READBACK_2:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static unsigned int wm8993_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *reg_cache = codec->reg_cache;
+
+ BUG_ON(reg > WM8993_MAX_REGISTER);
+
+ if (wm8993_volatile(reg))
+ return wm8993_read_hw(codec, reg);
+ else
+ return reg_cache[reg];
+}
+
+static int wm8993_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *reg_cache = codec->reg_cache;
+ u8 data[3];
+ int ret;
+
+ BUG_ON(reg > WM8993_MAX_REGISTER);
+
+ /* data is
+ * D15..D9 WM8993 register offset
+ * D8...D0 register data
+ */
+ data[0] = reg;
+ data[1] = value >> 8;
+ data[2] = value & 0x00ff;
+
+ if (!wm8993_volatile(reg))
+ reg_cache[reg] = value;
+
+ ret = codec->hw_write(codec->control_data, data, 3);
+
+ if (ret == 3)
+ return 0;
+ if (ret < 0)
+ return ret;
+ return -EIO;
+}
+
+struct _fll_div {
+ u16 fll_fratio;
+ u16 fll_outdiv;
+ u16 fll_clk_ref_div;
+ u16 n;
+ u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static struct {
+ unsigned int min;
+ unsigned int max;
+ u16 fll_fratio;
+ int ratio;
+} fll_fratios[] = {
+ { 0, 64000, 4, 16 },
+ { 64000, 128000, 3, 8 },
+ { 128000, 256000, 2, 4 },
+ { 256000, 1000000, 1, 2 },
+ { 1000000, 13500000, 0, 1 },
+};
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+ unsigned int Fout)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+ unsigned int div;
+ int i;
+
+ /* Fref must be <=13.5MHz */
+ div = 1;
+ fll_div->fll_clk_ref_div = 0;
+ while ((Fref / div) > 13500000) {
+ div *= 2;
+ fll_div->fll_clk_ref_div++;
+
+ if (div > 8) {
+ pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
+ }
+
+ pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
+
+ /* Apply the division for our remaining calculations */
+ Fref /= div;
+
+ /* Fvco should be 90-100MHz; don't check the upper bound */
+ div = 0;
+ target = Fout * 2;
+ while (target < 90000000) {
+ div++;
+ target *= 2;
+ if (div > 7) {
+ pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+ }
+ fll_div->fll_outdiv = div;
+
+ pr_debug("Fvco=%dHz\n", target);
+
+ /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ fll_div->fll_fratio = fll_fratios[i].fll_fratio;
+ target /= fll_fratios[i].ratio;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_fratios)) {
+ pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
+ return -EINVAL;
+ }
+
+ /* Now, calculate N.K */
+ Ndiv = target / Fref;
+
+ fll_div->n = Ndiv;
+ Nmod = target % Fref;
+ pr_debug("Nmod=%d\n", Nmod);
+
+ /* Calculate fractional part - scale up so we can round. */
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, Fref);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ fll_div->k = K / 10;
+
+ pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
+ fll_div->n, fll_div->k,
+ fll_div->fll_fratio, fll_div->fll_outdiv,
+ fll_div->fll_clk_ref_div);
+
+ return 0;
+}
+
+static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ u16 reg1, reg4, reg5;
+ struct _fll_div fll_div;
+ int ret;
+
+ /* Any change? */
+ if (Fref == wm8993->fll_fref && Fout == wm8993->fll_fout)
+ return 0;
+
+ /* Disable the FLL */
+ if (Fout == 0) {
+ dev_dbg(codec->dev, "FLL disabled\n");
+ wm8993->fll_fref = 0;
+ wm8993->fll_fout = 0;
+
+ reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1);
+ reg1 &= ~WM8993_FLL_ENA;
+ wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1);
+
+ return 0;
+ }
+
+ ret = fll_factors(&fll_div, Fref, Fout);
+ if (ret != 0)
+ return ret;
+
+ reg5 = wm8993_read(codec, WM8993_FLL_CONTROL_5);
+ reg5 &= ~WM8993_FLL_CLK_SRC_MASK;
+
+ switch (fll_id) {
+ case WM8993_FLL_MCLK:
+ break;
+
+ case WM8993_FLL_LRCLK:
+ reg5 |= 1;
+ break;
+
+ case WM8993_FLL_BCLK:
+ reg5 |= 2;
+ break;
+
+ default:
+ dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
+ return -EINVAL;
+ }
+
+ /* Any FLL configuration change requires that the FLL be
+ * disabled first. */
+ reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1);
+ reg1 &= ~WM8993_FLL_ENA;
+ wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1);
+
+ /* Apply the configuration */
+ if (fll_div.k)
+ reg1 |= WM8993_FLL_FRAC_MASK;
+ else
+ reg1 &= ~WM8993_FLL_FRAC_MASK;
+ wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1);
+
+ wm8993_write(codec, WM8993_FLL_CONTROL_2,
+ (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) |
+ (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT));
+ wm8993_write(codec, WM8993_FLL_CONTROL_3, fll_div.k);
+
+ reg4 = wm8993_read(codec, WM8993_FLL_CONTROL_4);
+ reg4 &= ~WM8993_FLL_N_MASK;
+ reg4 |= fll_div.n << WM8993_FLL_N_SHIFT;
+ wm8993_write(codec, WM8993_FLL_CONTROL_4, reg4);
+
+ reg5 &= ~WM8993_FLL_CLK_REF_DIV_MASK;
+ reg5 |= fll_div.fll_clk_ref_div << WM8993_FLL_CLK_REF_DIV_SHIFT;
+ wm8993_write(codec, WM8993_FLL_CONTROL_5, reg5);
+
+ /* Enable the FLL */
+ wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA);
+
+ dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
+
+ wm8993->fll_fref = Fref;
+ wm8993->fll_fout = Fout;
+
+ return 0;
+}
+
+static int configure_clock(struct snd_soc_codec *codec)
+{
+ struct wm8993_priv *wm8993 = codec->private_data;
+ unsigned int reg;
+
+ /* This should be done on init() for bypass paths */
+ switch (wm8993->sysclk_source) {
+ case WM8993_SYSCLK_MCLK:
+ dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8993->mclk_rate);
+
+ reg = wm8993_read(codec, WM8993_CLOCKING_2);
+ reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC);
+ if (wm8993->mclk_rate > 13500000) {
+ reg |= WM8993_MCLK_DIV;
+ wm8993->sysclk_rate = wm8993->mclk_rate / 2;
+ } else {
+ reg &= ~WM8993_MCLK_DIV;
+ wm8993->sysclk_rate = wm8993->mclk_rate;
+ }
+ wm8993_write(codec, WM8993_CLOCKING_2, reg);
+ break;
+
+ case WM8993_SYSCLK_FLL:
+ dev_dbg(codec->dev, "Using %dHz FLL clock\n",
+ wm8993->fll_fout);
+
+ reg = wm8993_read(codec, WM8993_CLOCKING_2);
+ reg |= WM8993_SYSCLK_SRC;
+ if (wm8993->fll_fout > 13500000) {
+ reg |= WM8993_MCLK_DIV;
+ wm8993->sysclk_rate = wm8993->fll_fout / 2;
+ } else {
+ reg &= ~WM8993_MCLK_DIV;
+ wm8993->sysclk_rate = wm8993->fll_fout;
+ }
+ wm8993_write(codec, WM8993_CLOCKING_2, reg);
+ break;
+
+ default:
+ dev_err(codec->dev, "System clock not configured\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm8993->sysclk_rate);
+
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
+static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
+static const unsigned int drc_max_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -1800, 300, 0);
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0);
+
+static const char *dac_deemph_text[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+ SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text);
+
+static const char *adc_hpf_text[] = {
+ "Hi-Fi",
+ "Voice 1",
+ "Voice 2",
+ "Voice 3",
+};
+
+static const struct soc_enum adc_hpf =
+ SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text);
+
+static const char *drc_path_text[] = {
+ "ADC",
+ "DAC"
+};
+
+static const struct soc_enum drc_path =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text);
+
+static const char *drc_r0_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "1/16",
+ "0",
+};
+
+static const struct soc_enum drc_r0 =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text);
+
+static const char *drc_r1_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "0",
+};
+
+static const struct soc_enum drc_r1 =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text);
+
+static const char *drc_attack_text[] = {
+ "Reserved",
+ "181us",
+ "363us",
+ "726us",
+ "1.45ms",
+ "2.9ms",
+ "5.8ms",
+ "11.6ms",
+ "23.2ms",
+ "46.4ms",
+ "92.8ms",
+ "185.6ms",
+};
+
+static const struct soc_enum drc_attack =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text);
+
+static const char *drc_decay_text[] = {
+ "186ms",
+ "372ms",
+ "743ms",
+ "1.49s",
+ "2.97ms",
+ "5.94ms",
+ "11.89ms",
+ "23.78ms",
+ "47.56ms",
+};
+
+static const struct soc_enum drc_decay =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text);
+
+static const char *drc_ff_text[] = {
+ "5 samples",
+ "9 samples",
+};
+
+static const struct soc_enum drc_ff =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text);
+
+static const char *drc_qr_rate_text[] = {
+ "0.725ms",
+ "1.45ms",
+ "5.8ms",
+};
+
+static const struct soc_enum drc_qr_rate =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text);
+
+static const char *drc_smooth_text[] = {
+ "Low",
+ "Medium",
+ "High",
+};
+
+static const struct soc_enum drc_smooth =
+ SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text);
+
+static const struct snd_kcontrol_new wm8993_snd_controls[] = {
+SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
+ 5, 9, 12, 0, sidetone_tlv),
+
+SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0),
+SOC_ENUM("DRC Path", drc_path),
+SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2,
+ 2, 60, 1, drc_comp_threash),
+SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3,
+ 11, 30, 1, drc_comp_amp),
+SOC_ENUM("DRC R0", drc_r0),
+SOC_ENUM("DRC R1", drc_r1),
+SOC_SINGLE_TLV("DRC Minimum Volume", WM8993_DRC_CONTROL_1, 2, 3, 1,
+ drc_min_tlv),
+SOC_SINGLE_TLV("DRC Maximum Volume", WM8993_DRC_CONTROL_1, 0, 3, 0,
+ drc_max_tlv),
+SOC_ENUM("DRC Attack Rate", drc_attack),
+SOC_ENUM("DRC Decay Rate", drc_decay),
+SOC_ENUM("DRC FF Delay", drc_ff),
+SOC_SINGLE("DRC Anti-clip Switch", WM8993_DRC_CONTROL_1, 9, 1, 0),
+SOC_SINGLE("DRC Quick Release Switch", WM8993_DRC_CONTROL_1, 10, 1, 0),
+SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0,
+ drc_qr_tlv),
+SOC_ENUM("DRC Quick Release Rate", drc_qr_rate),
+SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0),
+SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0),
+SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth),
+SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0,
+ drc_startup_tlv),
+
+SOC_SINGLE("EQ Switch", WM8993_EQ1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Volume", WM8993_LEFT_ADC_DIGITAL_VOLUME,
+ WM8993_RIGHT_ADC_DIGITAL_VOLUME, 1, 96, 0, digital_tlv),
+SOC_SINGLE("ADC High Pass Filter Switch", WM8993_ADC_CTRL, 8, 1, 0),
+SOC_ENUM("ADC High Pass Filter Mode", adc_hpf),
+
+SOC_DOUBLE_R_TLV("Playback Volume", WM8993_LEFT_DAC_DIGITAL_VOLUME,
+ WM8993_RIGHT_DAC_DIGITAL_VOLUME, 1, 96, 0, digital_tlv),
+SOC_SINGLE_TLV("Playback Boost Volume", WM8993_AUDIO_INTERFACE_2, 10, 3, 0,
+ dac_boost_tlv),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+
+SOC_SINGLE_TLV("SPKL DAC Volume", WM8993_SPKMIXL_ATTENUATION,
+ 2, 1, 1, wm_hubs_spkmix_tlv),
+
+SOC_SINGLE_TLV("SPKR DAC Volume", WM8993_SPKMIXR_ATTENUATION,
+ 2, 1, 1, wm_hubs_spkmix_tlv),
+};
+
+static const struct snd_kcontrol_new wm8993_eq_controls[] = {
+SOC_SINGLE_TLV("EQ1 Volume", WM8993_EQ2, 0, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 Volume", WM8993_EQ3, 0, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 Volume", WM8993_EQ4, 0, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 Volume", WM8993_EQ5, 0, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ5 Volume", WM8993_EQ6, 0, 24, 0, eq_tlv),
+};
+
+static int clk_sys_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return configure_clock(codec);
+
+ case SND_SOC_DAPM_POST_PMD:
+ break;
+ }
+
+ return 0;
+}
+
+/*
+ * When used with DAC outputs only the WM8993 charge pump supports
+ * operation in class W mode, providing very low power consumption
+ * when used with digital sources. Enable and disable this mode
+ * automatically depending on the mixer configuration.
+ *
+ * Currently the only supported paths are the direct DAC->headphone
+ * paths (which provide minimum power consumption anyway).
+ */
+static int class_w_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_codec *codec = widget->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ int ret;
+
+ /* Turn it off if we're using the main output mixer */
+ if (ucontrol->value.integer.value[0] == 0) {
+ if (wm8993->class_w_users == 0) {
+ dev_dbg(codec->dev, "Disabling Class W\n");
+ snd_soc_update_bits(codec, WM8993_CLASS_W_0,
+ WM8993_CP_DYN_FREQ |
+ WM8993_CP_DYN_V,
+ 0);
+ }
+ wm8993->class_w_users++;
+ }
+
+ /* Implement the change */
+ ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
+
+ /* Enable it if we're using the direct DAC path */
+ if (ucontrol->value.integer.value[0] == 1) {
+ if (wm8993->class_w_users == 1) {
+ dev_dbg(codec->dev, "Enabling Class W\n");
+ snd_soc_update_bits(codec, WM8993_CLASS_W_0,
+ WM8993_CP_DYN_FREQ |
+ WM8993_CP_DYN_V,
+ WM8993_CP_DYN_FREQ |
+ WM8993_CP_DYN_V);
+ }
+ wm8993->class_w_users--;
+ }
+
+ dev_dbg(codec->dev, "Indirect DAC use count now %d\n",
+ wm8993->class_w_users);
+
+ return ret;
+}
+
+#define SOC_DAPM_ENUM_W(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_double, \
+ .put = class_w_put, \
+ .private_value = (unsigned long)&xenum }
+
+static const char *hp_mux_text[] = {
+ "Mixer",
+ "DAC",
+};
+
+static const struct soc_enum hpl_enum =
+ SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
+
+static const struct snd_kcontrol_new hpl_mux =
+ SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum);
+
+static const struct soc_enum hpr_enum =
+ SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
+
+static const struct snd_kcontrol_new hpr_mux =
+ SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum);
+
+static const struct snd_kcontrol_new left_speaker_mixer[] = {
+SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0),
+SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Output Switch", WM8993_SPEAKER_MIXER, 3, 1, 0),
+SOC_DAPM_SINGLE("DAC Switch", WM8993_SPEAKER_MIXER, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_speaker_mixer[] = {
+SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 6, 1, 0),
+SOC_DAPM_SINGLE("IN1RP Switch", WM8993_SPEAKER_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Output Switch", WM8993_SPEAKER_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("DAC Switch", WM8993_SPEAKER_MIXER, 0, 1, 0),
+};
+
+static const char *aif_text[] = {
+ "Left", "Right"
+};
+
+static const struct soc_enum aifoutl_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text);
+
+static const struct snd_kcontrol_new aifoutl_mux =
+ SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
+
+static const struct soc_enum aifoutr_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text);
+
+static const struct snd_kcontrol_new aifoutr_mux =
+ SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
+
+static const struct soc_enum aifinl_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text);
+
+static const struct snd_kcontrol_new aifinl_mux =
+ SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
+
+static const struct soc_enum aifinr_enum =
+ SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text);
+
+static const struct snd_kcontrol_new aifinr_mux =
+ SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
+
+static const char *sidetone_text[] = {
+ "None", "Left", "Right"
+};
+
+static const struct soc_enum sidetonel_enum =
+ SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text);
+
+static const struct snd_kcontrol_new sidetonel_mux =
+ SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum);
+
+static const struct soc_enum sidetoner_enum =
+ SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text);
+
+static const struct snd_kcontrol_new sidetoner_mux =
+ SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum);
+
+static const struct snd_soc_dapm_widget wm8993_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0),
+
+SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0),
+
+SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux),
+SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux),
+
+SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux),
+SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux),
+
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &sidetonel_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux),
+
+SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
+
+SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0,
+ left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
+SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0,
+ right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
+
+};
+
+static const struct snd_soc_dapm_route routes[] = {
+ { "ADCL", NULL, "CLK_SYS" },
+ { "ADCL", NULL, "CLK_DSP" },
+ { "ADCR", NULL, "CLK_SYS" },
+ { "ADCR", NULL, "CLK_DSP" },
+
+ { "AIFOUTL Mux", "Left", "ADCL" },
+ { "AIFOUTL Mux", "Right", "ADCR" },
+ { "AIFOUTR Mux", "Left", "ADCL" },
+ { "AIFOUTR Mux", "Right", "ADCR" },
+
+ { "AIFOUTL", NULL, "AIFOUTL Mux" },
+ { "AIFOUTR", NULL, "AIFOUTR Mux" },
+
+ { "DACL Mux", "Left", "AIFINL" },
+ { "DACL Mux", "Right", "AIFINR" },
+ { "DACR Mux", "Left", "AIFINL" },
+ { "DACR Mux", "Right", "AIFINR" },
+
+ { "DACL Sidetone", "Left", "ADCL" },
+ { "DACL Sidetone", "Right", "ADCR" },
+ { "DACR Sidetone", "Left", "ADCL" },
+ { "DACR Sidetone", "Right", "ADCR" },
+
+ { "DACL", NULL, "CLK_SYS" },
+ { "DACL", NULL, "CLK_DSP" },
+ { "DACL", NULL, "DACL Mux" },
+ { "DACL", NULL, "DACL Sidetone" },
+ { "DACR", NULL, "CLK_SYS" },
+ { "DACR", NULL, "CLK_DSP" },
+ { "DACR", NULL, "DACR Mux" },
+ { "DACR", NULL, "DACR Sidetone" },
+
+ { "Left Output Mixer", "DAC Switch", "DACL" },
+
+ { "Right Output Mixer", "DAC Switch", "DACR" },
+
+ { "Left Output PGA", NULL, "CLK_SYS" },
+
+ { "Right Output PGA", NULL, "CLK_SYS" },
+
+ { "SPKL", "DAC Switch", "DACL" },
+ { "SPKL", NULL, "CLK_SYS" },
+
+ { "SPKR", "DAC Switch", "DACR" },
+ { "SPKR", NULL, "CLK_SYS" },
+
+ { "Left Headphone Mux", "DAC", "DACL" },
+ { "Right Headphone Mux", "DAC", "DACR" },
+};
+
+static int wm8993_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8993_priv *wm8993 = codec->private_data;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*40k */
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_VMID_SEL_MASK, 0x2);
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_2,
+ WM8993_TSHUT_ENA, WM8993_TSHUT_ENA);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Bring up VMID with fast soft start */
+ snd_soc_update_bits(codec, WM8993_ANTIPOP2,
+ WM8993_STARTUP_BIAS_ENA |
+ WM8993_VMID_BUF_ENA |
+ WM8993_VMID_RAMP_MASK |
+ WM8993_BIAS_SRC,
+ WM8993_STARTUP_BIAS_ENA |
+ WM8993_VMID_BUF_ENA |
+ WM8993_VMID_RAMP_MASK |
+ WM8993_BIAS_SRC);
+
+ /* If either line output is single ended we
+ * need the VMID buffer */
+ if (!wm8993->pdata.lineout1_diff ||
+ !wm8993->pdata.lineout2_diff)
+ snd_soc_update_bits(codec, WM8993_ANTIPOP1,
+ WM8993_LINEOUT_VMID_BUF_ENA,
+ WM8993_LINEOUT_VMID_BUF_ENA);
+
+ /* VMID=2*40k */
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_VMID_SEL_MASK |
+ WM8993_BIAS_ENA,
+ WM8993_BIAS_ENA | 0x2);
+ msleep(32);
+
+ /* Switch to normal bias */
+ snd_soc_update_bits(codec, WM8993_ANTIPOP2,
+ WM8993_BIAS_SRC |
+ WM8993_STARTUP_BIAS_ENA, 0);
+ }
+
+ /* VMID=2*240k */
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_VMID_SEL_MASK, 0x4);
+
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_2,
+ WM8993_TSHUT_ENA, 0);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, WM8993_ANTIPOP1,
+ WM8993_LINEOUT_VMID_BUF_ENA, 0);
+
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA,
+ 0);
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static int wm8993_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+
+ switch (clk_id) {
+ case WM8993_SYSCLK_MCLK:
+ wm8993->mclk_rate = freq;
+ case WM8993_SYSCLK_FLL:
+ wm8993->sysclk_source = clk_id;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8993_set_dai_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ unsigned int aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1);
+ unsigned int aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4);
+
+ aif1 &= ~(WM8993_BCLK_DIR | WM8993_AIF_BCLK_INV |
+ WM8993_AIF_LRCLK_INV | WM8993_AIF_FMT_MASK);
+ aif4 &= ~WM8993_LRCLK_DIR;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ wm8993->master = 0;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ aif4 |= WM8993_LRCLK_DIR;
+ wm8993->master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ aif1 |= WM8993_BCLK_DIR;
+ wm8993->master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif1 |= WM8993_BCLK_DIR;
+ aif4 |= WM8993_LRCLK_DIR;
+ wm8993->master = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_B:
+ aif1 |= WM8993_AIF_LRCLK_INV;
+ case SND_SOC_DAIFMT_DSP_A:
+ aif1 |= 0x18;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ aif1 |= 0x10;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif1 |= 0x8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8993_AIF_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif1 |= WM8993_AIF_BCLK_INV | WM8993_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8993_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif1 |= WM8993_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1);
+ wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4);
+
+ return 0;
+}
+
+static int wm8993_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ int ret, i, best, best_val, cur_val;
+ unsigned int clocking1, clocking3, aif1, aif4;
+
+ clocking1 = wm8993_read(codec, WM8993_CLOCKING_1);
+ clocking1 &= ~WM8993_BCLK_DIV_MASK;
+
+ clocking3 = wm8993_read(codec, WM8993_CLOCKING_3);
+ clocking3 &= ~(WM8993_CLK_SYS_RATE_MASK | WM8993_SAMPLE_RATE_MASK);
+
+ aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1);
+ aif1 &= ~WM8993_AIF_WL_MASK;
+
+ aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4);
+ aif4 &= ~WM8993_LRCLK_RATE_MASK;
+
+ /* What BCLK do we need? */
+ wm8993->fs = params_rate(params);
+ wm8993->bclk = 2 * wm8993->fs;
+ if (wm8993->tdm_slots) {
+ dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n",
+ wm8993->tdm_slots, wm8993->tdm_width);
+ wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots;
+ } else {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ wm8993->bclk *= 16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ wm8993->bclk *= 20;
+ aif1 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ wm8993->bclk *= 24;
+ aif1 |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ wm8993->bclk *= 32;
+ aif1 |= 0x18;
+ break;
+ default:
+ return -EINVAL;
+ }
+ }
+
+ dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8993->bclk);
+
+ ret = configure_clock(codec);
+ if (ret != 0)
+ return ret;
+
+ /* Select nearest CLK_SYS_RATE */
+ best = 0;
+ best_val = abs((wm8993->sysclk_rate / clk_sys_rates[0].ratio)
+ - wm8993->fs);
+ for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
+ cur_val = abs((wm8993->sysclk_rate /
+ clk_sys_rates[i].ratio) - wm8993->fs);;
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n",
+ clk_sys_rates[best].ratio);
+ clocking3 |= (clk_sys_rates[best].clk_sys_rate
+ << WM8993_CLK_SYS_RATE_SHIFT);
+
+ /* SAMPLE_RATE */
+ best = 0;
+ best_val = abs(wm8993->fs - sample_rates[0].rate);
+ for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+ /* Closest match */
+ cur_val = abs(wm8993->fs - sample_rates[i].rate);
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n",
+ sample_rates[best].rate);
+ clocking3 |= (sample_rates[best].sample_rate
+ << WM8993_SAMPLE_RATE_SHIFT);
+
+ /* BCLK_DIV */
+ best = 0;
+ best_val = INT_MAX;
+ for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+ cur_val = ((wm8993->sysclk_rate * 10) / bclk_divs[i].div)
+ - wm8993->bclk;
+ if (cur_val < 0) /* Table is sorted */
+ break;
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ wm8993->bclk = (wm8993->sysclk_rate * 10) / bclk_divs[best].div;
+ dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n",
+ bclk_divs[best].div, wm8993->bclk);
+ clocking1 |= bclk_divs[best].bclk_div << WM8993_BCLK_DIV_SHIFT;
+
+ /* LRCLK is a simple fraction of BCLK */
+ dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8993->bclk / wm8993->fs);
+ aif4 |= wm8993->bclk / wm8993->fs;
+
+ wm8993_write(codec, WM8993_CLOCKING_1, clocking1);
+ wm8993_write(codec, WM8993_CLOCKING_3, clocking3);
+ wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1);
+ wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4);
+
+ /* ReTune Mobile? */
+ if (wm8993->pdata.num_retune_configs) {
+ u16 eq1 = wm8993_read(codec, WM8993_EQ1);
+ struct wm8993_retune_mobile_setting *s;
+
+ best = 0;
+ best_val = abs(wm8993->pdata.retune_configs[0].rate
+ - wm8993->fs);
+ for (i = 0; i < wm8993->pdata.num_retune_configs; i++) {
+ cur_val = abs(wm8993->pdata.retune_configs[i].rate
+ - wm8993->fs);
+ if (cur_val < best_val) {
+ best_val = cur_val;
+ best = i;
+ }
+ }
+ s = &wm8993->pdata.retune_configs[best];
+
+ dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
+ s->name, s->rate);
+
+ /* Disable EQ while we reconfigure */
+ snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, 0);
+
+ for (i = 1; i < ARRAY_SIZE(s->config); i++)
+ wm8993_write(codec, WM8993_EQ1 + i, s->config[i]);
+
+ snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, eq1);
+ }
+
+ return 0;
+}
+
+static int wm8993_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int reg;
+
+ reg = wm8993_read(codec, WM8993_DAC_CTRL);
+
+ if (mute)
+ reg |= WM8993_DAC_MUTE;
+ else
+ reg &= ~WM8993_DAC_MUTE;
+
+ wm8993_write(codec, WM8993_DAC_CTRL, reg);
+
+ return 0;
+}
+
+static int wm8993_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8993_priv *wm8993 = codec->private_data;
+ int aif1 = 0;
+ int aif2 = 0;
+
+ /* Don't need to validate anything if we're turning off TDM */
+ if (slots == 0) {
+ wm8993->tdm_slots = 0;
+ goto out;
+ }
+
+ /* Note that we allow configurations we can't handle ourselves -
+ * for example, we can generate clocks for slots 2 and up even if
+ * we can't use those slots ourselves.
+ */
+ aif1 |= WM8993_AIFADC_TDM;
+ aif2 |= WM8993_AIFDAC_TDM;
+
+ switch (rx_mask) {
+ case 3:
+ break;
+ case 0xc:
+ aif1 |= WM8993_AIFADC_TDM_CHAN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+
+ switch (tx_mask) {
+ case 3:
+ break;
+ case 0xc:
+ aif2 |= WM8993_AIFDAC_TDM_CHAN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+out:
+ wm8993->tdm_width = slot_width;
+ wm8993->tdm_slots = slots / 2;
+
+ snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_1,
+ WM8993_AIFADC_TDM | WM8993_AIFADC_TDM_CHAN, aif1);
+ snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_2,
+ WM8993_AIFDAC_TDM | WM8993_AIFDAC_TDM_CHAN, aif2);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops wm8993_ops = {
+ .set_sysclk = wm8993_set_sysclk,
+ .set_fmt = wm8993_set_dai_fmt,
+ .hw_params = wm8993_hw_params,
+ .digital_mute = wm8993_digital_mute,
+ .set_pll = wm8993_set_fll,
+ .set_tdm_slot = wm8993_set_tdm_slot,
+};
+
+#define WM8993_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8993_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai wm8993_dai = {
+ .name = "WM8993",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8993_RATES,
+ .formats = WM8993_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8993_RATES,
+ .formats = WM8993_FORMATS,
+ },
+ .ops = &wm8993_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8993_dai);
+
+static struct snd_soc_codec *wm8993_codec;
+
+static int wm8993_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct wm8993_priv *wm8993;
+ int ret = 0;
+
+ if (!wm8993_codec) {
+ dev_err(&pdev->dev, "I2C device not yet probed\n");
+ goto err;
+ }
+
+ socdev->card->codec = wm8993_codec;
+ codec = wm8993_codec;
+ wm8993 = codec->private_data;
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms\n");
+ goto err;
+ }
+
+ snd_soc_add_controls(codec, wm8993_snd_controls,
+ ARRAY_SIZE(wm8993_snd_controls));
+ if (wm8993->pdata.num_retune_configs != 0) {
+ dev_dbg(codec->dev, "Using ReTune Mobile\n");
+ } else {
+ dev_dbg(codec->dev, "No ReTune Mobile, using normal EQ\n");
+ snd_soc_add_controls(codec, wm8993_eq_controls,
+ ARRAY_SIZE(wm8993_eq_controls));
+ }
+
+ snd_soc_dapm_new_controls(codec, wm8993_dapm_widgets,
+ ARRAY_SIZE(wm8993_dapm_widgets));
+ wm_hubs_add_analogue_controls(codec);
+
+ snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+ wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff,
+ wm8993->pdata.lineout2_diff);
+
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+err:
+ return ret;
+}
+
+static int wm8993_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8993 = {
+ .probe = wm8993_probe,
+ .remove = wm8993_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8993);
+
+static int wm8993_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8993_priv *wm8993;
+ struct snd_soc_codec *codec;
+ unsigned int val;
+ int ret;
+
+ if (wm8993_codec) {
+ dev_err(&i2c->dev, "A WM8993 is already registered\n");
+ return -EINVAL;
+ }
+
+ wm8993 = kzalloc(sizeof(struct wm8993_priv), GFP_KERNEL);
+ if (wm8993 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8993->codec;
+ if (i2c->dev.platform_data)
+ memcpy(&wm8993->pdata, i2c->dev.platform_data,
+ sizeof(wm8993->pdata));
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->name = "WM8993";
+ codec->read = wm8993_read;
+ codec->write = wm8993_write;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->reg_cache = wm8993->reg_cache;
+ codec->reg_cache_size = ARRAY_SIZE(wm8993->reg_cache);
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8993_set_bias_level;
+ codec->dai = &wm8993_dai;
+ codec->num_dai = 1;
+ codec->private_data = wm8993;
+
+ memcpy(wm8993->reg_cache, wm8993_reg_defaults,
+ sizeof(wm8993->reg_cache));
+
+ i2c_set_clientdata(i2c, wm8993);
+ codec->control_data = i2c;
+ wm8993_codec = codec;
+
+ codec->dev = &i2c->dev;
+
+ val = wm8993_read_hw(codec, WM8993_SOFTWARE_RESET);
+ if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) {
+ dev_err(codec->dev, "Invalid ID register value %x\n", val);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = wm8993_write(codec, WM8993_SOFTWARE_RESET, 0xffff);
+ if (ret != 0)
+ goto err;
+
+ /* By default we're using the output mixers */
+ wm8993->class_w_users = 2;
+
+ /* Latch volume update bits and default ZC on */
+ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME,
+ WM8993_DAC_VU, WM8993_DAC_VU);
+ snd_soc_update_bits(codec, WM8993_RIGHT_ADC_DIGITAL_VOLUME,
+ WM8993_ADC_VU, WM8993_ADC_VU);
+
+ /* Manualy manage the HPOUT sequencing for independent stereo
+ * control. */
+ snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
+ WM8993_HPOUT1_AUTO_PU, 0);
+
+ /* Use automatic clock configuration */
+ snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0);
+
+ if (!wm8993->pdata.lineout1_diff)
+ snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
+ WM8993_LINEOUT1_MODE,
+ WM8993_LINEOUT1_MODE);
+ if (!wm8993->pdata.lineout2_diff)
+ snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
+ WM8993_LINEOUT2_MODE,
+ WM8993_LINEOUT2_MODE);
+
+ if (wm8993->pdata.lineout1fb)
+ snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+ WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
+
+ if (wm8993->pdata.lineout2fb)
+ snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+ WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
+
+ /* Apply the microphone bias/detection configuration - the
+ * platform data is directly applicable to the register. */
+ snd_soc_update_bits(codec, WM8993_MICBIAS,
+ WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
+ WM8993_MICB1_LVL | WM8993_MICB2_LVL,
+ wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT |
+ wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT |
+ wm8993->pdata.micbias1_lvl |
+ wm8993->pdata.micbias1_lvl << 1);
+
+ ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (ret != 0)
+ goto err;
+
+ wm8993_dai.dev = codec->dev;
+
+ ret = snd_soc_register_dai(&wm8993_dai);
+ if (ret != 0)
+ goto err_bias;
+
+ ret = snd_soc_register_codec(codec);
+
+ return 0;
+
+err_bias:
+ wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
+err:
+ wm8993_codec = NULL;
+ kfree(wm8993);
+ return ret;
+}
+
+static int wm8993_i2c_remove(struct i2c_client *client)
+{
+ struct wm8993_priv *wm8993 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&wm8993->codec);
+ snd_soc_unregister_dai(&wm8993_dai);
+
+ wm8993_set_bias_level(&wm8993->codec, SND_SOC_BIAS_OFF);
+ kfree(wm8993);
+
+ return 0;
+}
+
+static const struct i2c_device_id wm8993_i2c_id[] = {
+ { "wm8993", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id);
+
+static struct i2c_driver wm8993_i2c_driver = {
+ .driver = {
+ .name = "WM8993",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8993_i2c_probe,
+ .remove = wm8993_i2c_remove,
+ .id_table = wm8993_i2c_id,
+};
+
+
+static int __init wm8993_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8993_i2c_driver);
+ if (ret != 0)
+ pr_err("WM8993: Unable to register I2C driver: %d\n", ret);
+
+ return ret;
+}
+module_init(wm8993_modinit);
+
+static void __exit wm8993_exit(void)
+{
+ i2c_del_driver(&wm8993_i2c_driver);
+}
+module_exit(wm8993_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8993 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8993.h b/sound/soc/codecs/wm8993.h
new file mode 100644
index 0000000..30e71ca
--- /dev/null
+++ b/sound/soc/codecs/wm8993.h
@@ -0,0 +1,2132 @@
+#ifndef WM8993_H
+#define WM8993_H
+
+extern struct snd_soc_dai wm8993_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8993;
+
+#define WM8993_SYSCLK_MCLK 1
+#define WM8993_SYSCLK_FLL 2
+
+#define WM8993_FLL_MCLK 1
+#define WM8993_FLL_BCLK 2
+#define WM8993_FLL_LRCLK 3
+
+/*
+ * Register values.
+ */
+#define WM8993_SOFTWARE_RESET 0x00
+#define WM8993_POWER_MANAGEMENT_1 0x01
+#define WM8993_POWER_MANAGEMENT_2 0x02
+#define WM8993_POWER_MANAGEMENT_3 0x03
+#define WM8993_AUDIO_INTERFACE_1 0x04
+#define WM8993_AUDIO_INTERFACE_2 0x05
+#define WM8993_CLOCKING_1 0x06
+#define WM8993_CLOCKING_2 0x07
+#define WM8993_AUDIO_INTERFACE_3 0x08
+#define WM8993_AUDIO_INTERFACE_4 0x09
+#define WM8993_DAC_CTRL 0x0A
+#define WM8993_LEFT_DAC_DIGITAL_VOLUME 0x0B
+#define WM8993_RIGHT_DAC_DIGITAL_VOLUME 0x0C
+#define WM8993_DIGITAL_SIDE_TONE 0x0D
+#define WM8993_ADC_CTRL 0x0E
+#define WM8993_LEFT_ADC_DIGITAL_VOLUME 0x0F
+#define WM8993_RIGHT_ADC_DIGITAL_VOLUME 0x10
+#define WM8993_GPIO_CTRL_1 0x12
+#define WM8993_GPIO1 0x13
+#define WM8993_IRQ_DEBOUNCE 0x14
+#define WM8993_GPIOCTRL_2 0x16
+#define WM8993_GPIO_POL 0x17
+#define WM8993_LEFT_LINE_INPUT_1_2_VOLUME 0x18
+#define WM8993_LEFT_LINE_INPUT_3_4_VOLUME 0x19
+#define WM8993_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A
+#define WM8993_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B
+#define WM8993_LEFT_OUTPUT_VOLUME 0x1C
+#define WM8993_RIGHT_OUTPUT_VOLUME 0x1D
+#define WM8993_LINE_OUTPUTS_VOLUME 0x1E
+#define WM8993_HPOUT2_VOLUME 0x1F
+#define WM8993_LEFT_OPGA_VOLUME 0x20
+#define WM8993_RIGHT_OPGA_VOLUME 0x21
+#define WM8993_SPKMIXL_ATTENUATION 0x22
+#define WM8993_SPKMIXR_ATTENUATION 0x23
+#define WM8993_SPKOUT_MIXERS 0x24
+#define WM8993_SPKOUT_BOOST 0x25
+#define WM8993_SPEAKER_VOLUME_LEFT 0x26
+#define WM8993_SPEAKER_VOLUME_RIGHT 0x27
+#define WM8993_INPUT_MIXER2 0x28
+#define WM8993_INPUT_MIXER3 0x29
+#define WM8993_INPUT_MIXER4 0x2A
+#define WM8993_INPUT_MIXER5 0x2B
+#define WM8993_INPUT_MIXER6 0x2C
+#define WM8993_OUTPUT_MIXER1 0x2D
+#define WM8993_OUTPUT_MIXER2 0x2E
+#define WM8993_OUTPUT_MIXER3 0x2F
+#define WM8993_OUTPUT_MIXER4 0x30
+#define WM8993_OUTPUT_MIXER5 0x31
+#define WM8993_OUTPUT_MIXER6 0x32
+#define WM8993_HPOUT2_MIXER 0x33
+#define WM8993_LINE_MIXER1 0x34
+#define WM8993_LINE_MIXER2 0x35
+#define WM8993_SPEAKER_MIXER 0x36
+#define WM8993_ADDITIONAL_CONTROL 0x37
+#define WM8993_ANTIPOP1 0x38
+#define WM8993_ANTIPOP2 0x39
+#define WM8993_MICBIAS 0x3A
+#define WM8993_FLL_CONTROL_1 0x3C
+#define WM8993_FLL_CONTROL_2 0x3D
+#define WM8993_FLL_CONTROL_3 0x3E
+#define WM8993_FLL_CONTROL_4 0x3F
+#define WM8993_FLL_CONTROL_5 0x40
+#define WM8993_CLOCKING_3 0x41
+#define WM8993_CLOCKING_4 0x42
+#define WM8993_MW_SLAVE_CONTROL 0x43
+#define WM8993_BUS_CONTROL_1 0x45
+#define WM8993_WRITE_SEQUENCER_0 0x46
+#define WM8993_WRITE_SEQUENCER_1 0x47
+#define WM8993_WRITE_SEQUENCER_2 0x48
+#define WM8993_WRITE_SEQUENCER_3 0x49
+#define WM8993_WRITE_SEQUENCER_4 0x4A
+#define WM8993_WRITE_SEQUENCER_5 0x4B
+#define WM8993_CHARGE_PUMP_1 0x4C
+#define WM8993_CLASS_W_0 0x51
+#define WM8993_DC_SERVO_0 0x54
+#define WM8993_DC_SERVO_1 0x55
+#define WM8993_DC_SERVO_3 0x57
+#define WM8993_DC_SERVO_READBACK_0 0x58
+#define WM8993_DC_SERVO_READBACK_1 0x59
+#define WM8993_DC_SERVO_READBACK_2 0x5A
+#define WM8993_ANALOGUE_HP_0 0x60
+#define WM8993_EQ1 0x62
+#define WM8993_EQ2 0x63
+#define WM8993_EQ3 0x64
+#define WM8993_EQ4 0x65
+#define WM8993_EQ5 0x66
+#define WM8993_EQ6 0x67
+#define WM8993_EQ7 0x68
+#define WM8993_EQ8 0x69
+#define WM8993_EQ9 0x6A
+#define WM8993_EQ10 0x6B
+#define WM8993_EQ11 0x6C
+#define WM8993_EQ12 0x6D
+#define WM8993_EQ13 0x6E
+#define WM8993_EQ14 0x6F
+#define WM8993_EQ15 0x70
+#define WM8993_EQ16 0x71
+#define WM8993_EQ17 0x72
+#define WM8993_EQ18 0x73
+#define WM8993_EQ19 0x74
+#define WM8993_EQ20 0x75
+#define WM8993_EQ21 0x76
+#define WM8993_EQ22 0x77
+#define WM8993_EQ23 0x78
+#define WM8993_EQ24 0x79
+#define WM8993_DIGITAL_PULLS 0x7A
+#define WM8993_DRC_CONTROL_1 0x7B
+#define WM8993_DRC_CONTROL_2 0x7C
+#define WM8993_DRC_CONTROL_3 0x7D
+#define WM8993_DRC_CONTROL_4 0x7E
+
+#define WM8993_REGISTER_COUNT 0x7F
+#define WM8993_MAX_REGISTER 0x7E
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM8993_SW_RESET_MASK 0xFFFF /* SW_RESET - [15:0] */
+#define WM8993_SW_RESET_SHIFT 0 /* SW_RESET - [15:0] */
+#define WM8993_SW_RESET_WIDTH 16 /* SW_RESET - [15:0] */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM8993_SPKOUTR_ENA 0x2000 /* SPKOUTR_ENA */
+#define WM8993_SPKOUTR_ENA_MASK 0x2000 /* SPKOUTR_ENA */
+#define WM8993_SPKOUTR_ENA_SHIFT 13 /* SPKOUTR_ENA */
+#define WM8993_SPKOUTR_ENA_WIDTH 1 /* SPKOUTR_ENA */
+#define WM8993_SPKOUTL_ENA 0x1000 /* SPKOUTL_ENA */
+#define WM8993_SPKOUTL_ENA_MASK 0x1000 /* SPKOUTL_ENA */
+#define WM8993_SPKOUTL_ENA_SHIFT 12 /* SPKOUTL_ENA */
+#define WM8993_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */
+#define WM8993_HPOUT2_ENA 0x0800 /* HPOUT2_ENA */
+#define WM8993_HPOUT2_ENA_MASK 0x0800 /* HPOUT2_ENA */
+#define WM8993_HPOUT2_ENA_SHIFT 11 /* HPOUT2_ENA */
+#define WM8993_HPOUT2_ENA_WIDTH 1 /* HPOUT2_ENA */
+#define WM8993_HPOUT1L_ENA 0x0200 /* HPOUT1L_ENA */
+#define WM8993_HPOUT1L_ENA_MASK 0x0200 /* HPOUT1L_ENA */
+#define WM8993_HPOUT1L_ENA_SHIFT 9 /* HPOUT1L_ENA */
+#define WM8993_HPOUT1L_ENA_WIDTH 1 /* HPOUT1L_ENA */
+#define WM8993_HPOUT1R_ENA 0x0100 /* HPOUT1R_ENA */
+#define WM8993_HPOUT1R_ENA_MASK 0x0100 /* HPOUT1R_ENA */
+#define WM8993_HPOUT1R_ENA_SHIFT 8 /* HPOUT1R_ENA */
+#define WM8993_HPOUT1R_ENA_WIDTH 1 /* HPOUT1R_ENA */
+#define WM8993_MICB2_ENA 0x0020 /* MICB2_ENA */
+#define WM8993_MICB2_ENA_MASK 0x0020 /* MICB2_ENA */
+#define WM8993_MICB2_ENA_SHIFT 5 /* MICB2_ENA */
+#define WM8993_MICB2_ENA_WIDTH 1 /* MICB2_ENA */
+#define WM8993_MICB1_ENA 0x0010 /* MICB1_ENA */
+#define WM8993_MICB1_ENA_MASK 0x0010 /* MICB1_ENA */
+#define WM8993_MICB1_ENA_SHIFT 4 /* MICB1_ENA */
+#define WM8993_MICB1_ENA_WIDTH 1 /* MICB1_ENA */
+#define WM8993_VMID_SEL_MASK 0x0006 /* VMID_SEL - [2:1] */
+#define WM8993_VMID_SEL_SHIFT 1 /* VMID_SEL - [2:1] */
+#define WM8993_VMID_SEL_WIDTH 2 /* VMID_SEL - [2:1] */
+#define WM8993_BIAS_ENA 0x0001 /* BIAS_ENA */
+#define WM8993_BIAS_ENA_MASK 0x0001 /* BIAS_ENA */
+#define WM8993_BIAS_ENA_SHIFT 0 /* BIAS_ENA */
+#define WM8993_BIAS_ENA_WIDTH 1 /* BIAS_ENA */
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM8993_TSHUT_ENA 0x4000 /* TSHUT_ENA */
+#define WM8993_TSHUT_ENA_MASK 0x4000 /* TSHUT_ENA */
+#define WM8993_TSHUT_ENA_SHIFT 14 /* TSHUT_ENA */
+#define WM8993_TSHUT_ENA_WIDTH 1 /* TSHUT_ENA */
+#define WM8993_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */
+#define WM8993_TSHUT_OPDIS_MASK 0x2000 /* TSHUT_OPDIS */
+#define WM8993_TSHUT_OPDIS_SHIFT 13 /* TSHUT_OPDIS */
+#define WM8993_TSHUT_OPDIS_WIDTH 1 /* TSHUT_OPDIS */
+#define WM8993_OPCLK_ENA 0x0800 /* OPCLK_ENA */
+#define WM8993_OPCLK_ENA_MASK 0x0800 /* OPCLK_ENA */
+#define WM8993_OPCLK_ENA_SHIFT 11 /* OPCLK_ENA */
+#define WM8993_OPCLK_ENA_WIDTH 1 /* OPCLK_ENA */
+#define WM8993_MIXINL_ENA 0x0200 /* MIXINL_ENA */
+#define WM8993_MIXINL_ENA_MASK 0x0200 /* MIXINL_ENA */
+#define WM8993_MIXINL_ENA_SHIFT 9 /* MIXINL_ENA */
+#define WM8993_MIXINL_ENA_WIDTH 1 /* MIXINL_ENA */
+#define WM8993_MIXINR_ENA 0x0100 /* MIXINR_ENA */
+#define WM8993_MIXINR_ENA_MASK 0x0100 /* MIXINR_ENA */
+#define WM8993_MIXINR_ENA_SHIFT 8 /* MIXINR_ENA */
+#define WM8993_MIXINR_ENA_WIDTH 1 /* MIXINR_ENA */
+#define WM8993_IN2L_ENA 0x0080 /* IN2L_ENA */
+#define WM8993_IN2L_ENA_MASK 0x0080 /* IN2L_ENA */
+#define WM8993_IN2L_ENA_SHIFT 7 /* IN2L_ENA */
+#define WM8993_IN2L_ENA_WIDTH 1 /* IN2L_ENA */
+#define WM8993_IN1L_ENA 0x0040 /* IN1L_ENA */
+#define WM8993_IN1L_ENA_MASK 0x0040 /* IN1L_ENA */
+#define WM8993_IN1L_ENA_SHIFT 6 /* IN1L_ENA */
+#define WM8993_IN1L_ENA_WIDTH 1 /* IN1L_ENA */
+#define WM8993_IN2R_ENA 0x0020 /* IN2R_ENA */
+#define WM8993_IN2R_ENA_MASK 0x0020 /* IN2R_ENA */
+#define WM8993_IN2R_ENA_SHIFT 5 /* IN2R_ENA */
+#define WM8993_IN2R_ENA_WIDTH 1 /* IN2R_ENA */
+#define WM8993_IN1R_ENA 0x0010 /* IN1R_ENA */
+#define WM8993_IN1R_ENA_MASK 0x0010 /* IN1R_ENA */
+#define WM8993_IN1R_ENA_SHIFT 4 /* IN1R_ENA */
+#define WM8993_IN1R_ENA_WIDTH 1 /* IN1R_ENA */
+#define WM8993_ADCL_ENA 0x0002 /* ADCL_ENA */
+#define WM8993_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */
+#define WM8993_ADCL_ENA_SHIFT 1 /* ADCL_ENA */
+#define WM8993_ADCL_ENA_WIDTH 1 /* ADCL_ENA */
+#define WM8993_ADCR_ENA 0x0001 /* ADCR_ENA */
+#define WM8993_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */
+#define WM8993_ADCR_ENA_SHIFT 0 /* ADCR_ENA */
+#define WM8993_ADCR_ENA_WIDTH 1 /* ADCR_ENA */
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM8993_LINEOUT1N_ENA 0x2000 /* LINEOUT1N_ENA */
+#define WM8993_LINEOUT1N_ENA_MASK 0x2000 /* LINEOUT1N_ENA */
+#define WM8993_LINEOUT1N_ENA_SHIFT 13 /* LINEOUT1N_ENA */
+#define WM8993_LINEOUT1N_ENA_WIDTH 1 /* LINEOUT1N_ENA */
+#define WM8993_LINEOUT1P_ENA 0x1000 /* LINEOUT1P_ENA */
+#define WM8993_LINEOUT1P_ENA_MASK 0x1000 /* LINEOUT1P_ENA */
+#define WM8993_LINEOUT1P_ENA_SHIFT 12 /* LINEOUT1P_ENA */
+#define WM8993_LINEOUT1P_ENA_WIDTH 1 /* LINEOUT1P_ENA */
+#define WM8993_LINEOUT2N_ENA 0x0800 /* LINEOUT2N_ENA */
+#define WM8993_LINEOUT2N_ENA_MASK 0x0800 /* LINEOUT2N_ENA */
+#define WM8993_LINEOUT2N_ENA_SHIFT 11 /* LINEOUT2N_ENA */
+#define WM8993_LINEOUT2N_ENA_WIDTH 1 /* LINEOUT2N_ENA */
+#define WM8993_LINEOUT2P_ENA 0x0400 /* LINEOUT2P_ENA */
+#define WM8993_LINEOUT2P_ENA_MASK 0x0400 /* LINEOUT2P_ENA */
+#define WM8993_LINEOUT2P_ENA_SHIFT 10 /* LINEOUT2P_ENA */
+#define WM8993_LINEOUT2P_ENA_WIDTH 1 /* LINEOUT2P_ENA */
+#define WM8993_SPKRVOL_ENA 0x0200 /* SPKRVOL_ENA */
+#define WM8993_SPKRVOL_ENA_MASK 0x0200 /* SPKRVOL_ENA */
+#define WM8993_SPKRVOL_ENA_SHIFT 9 /* SPKRVOL_ENA */
+#define WM8993_SPKRVOL_ENA_WIDTH 1 /* SPKRVOL_ENA */
+#define WM8993_SPKLVOL_ENA 0x0100 /* SPKLVOL_ENA */
+#define WM8993_SPKLVOL_ENA_MASK 0x0100 /* SPKLVOL_ENA */
+#define WM8993_SPKLVOL_ENA_SHIFT 8 /* SPKLVOL_ENA */
+#define WM8993_SPKLVOL_ENA_WIDTH 1 /* SPKLVOL_ENA */
+#define WM8993_MIXOUTLVOL_ENA 0x0080 /* MIXOUTLVOL_ENA */
+#define WM8993_MIXOUTLVOL_ENA_MASK 0x0080 /* MIXOUTLVOL_ENA */
+#define WM8993_MIXOUTLVOL_ENA_SHIFT 7 /* MIXOUTLVOL_ENA */
+#define WM8993_MIXOUTLVOL_ENA_WIDTH 1 /* MIXOUTLVOL_ENA */
+#define WM8993_MIXOUTRVOL_ENA 0x0040 /* MIXOUTRVOL_ENA */
+#define WM8993_MIXOUTRVOL_ENA_MASK 0x0040 /* MIXOUTRVOL_ENA */
+#define WM8993_MIXOUTRVOL_ENA_SHIFT 6 /* MIXOUTRVOL_ENA */
+#define WM8993_MIXOUTRVOL_ENA_WIDTH 1 /* MIXOUTRVOL_ENA */
+#define WM8993_MIXOUTL_ENA 0x0020 /* MIXOUTL_ENA */
+#define WM8993_MIXOUTL_ENA_MASK 0x0020 /* MIXOUTL_ENA */
+#define WM8993_MIXOUTL_ENA_SHIFT 5 /* MIXOUTL_ENA */
+#define WM8993_MIXOUTL_ENA_WIDTH 1 /* MIXOUTL_ENA */
+#define WM8993_MIXOUTR_ENA 0x0010 /* MIXOUTR_ENA */
+#define WM8993_MIXOUTR_ENA_MASK 0x0010 /* MIXOUTR_ENA */
+#define WM8993_MIXOUTR_ENA_SHIFT 4 /* MIXOUTR_ENA */
+#define WM8993_MIXOUTR_ENA_WIDTH 1 /* MIXOUTR_ENA */
+#define WM8993_DACL_ENA 0x0002 /* DACL_ENA */
+#define WM8993_DACL_ENA_MASK 0x0002 /* DACL_ENA */
+#define WM8993_DACL_ENA_SHIFT 1 /* DACL_ENA */
+#define WM8993_DACL_ENA_WIDTH 1 /* DACL_ENA */
+#define WM8993_DACR_ENA 0x0001 /* DACR_ENA */
+#define WM8993_DACR_ENA_MASK 0x0001 /* DACR_ENA */
+#define WM8993_DACR_ENA_SHIFT 0 /* DACR_ENA */
+#define WM8993_DACR_ENA_WIDTH 1 /* DACR_ENA */
+
+/*
+ * R4 (0x04) - Audio Interface (1)
+ */
+#define WM8993_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */
+#define WM8993_AIFADCL_SRC_MASK 0x8000 /* AIFADCL_SRC */
+#define WM8993_AIFADCL_SRC_SHIFT 15 /* AIFADCL_SRC */
+#define WM8993_AIFADCL_SRC_WIDTH 1 /* AIFADCL_SRC */
+#define WM8993_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */
+#define WM8993_AIFADCR_SRC_MASK 0x4000 /* AIFADCR_SRC */
+#define WM8993_AIFADCR_SRC_SHIFT 14 /* AIFADCR_SRC */
+#define WM8993_AIFADCR_SRC_WIDTH 1 /* AIFADCR_SRC */
+#define WM8993_AIFADC_TDM 0x2000 /* AIFADC_TDM */
+#define WM8993_AIFADC_TDM_MASK 0x2000 /* AIFADC_TDM */
+#define WM8993_AIFADC_TDM_SHIFT 13 /* AIFADC_TDM */
+#define WM8993_AIFADC_TDM_WIDTH 1 /* AIFADC_TDM */
+#define WM8993_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */
+#define WM8993_AIFADC_TDM_CHAN_MASK 0x1000 /* AIFADC_TDM_CHAN */
+#define WM8993_AIFADC_TDM_CHAN_SHIFT 12 /* AIFADC_TDM_CHAN */
+#define WM8993_AIFADC_TDM_CHAN_WIDTH 1 /* AIFADC_TDM_CHAN */
+#define WM8993_BCLK_DIR 0x0200 /* BCLK_DIR */
+#define WM8993_BCLK_DIR_MASK 0x0200 /* BCLK_DIR */
+#define WM8993_BCLK_DIR_SHIFT 9 /* BCLK_DIR */
+#define WM8993_BCLK_DIR_WIDTH 1 /* BCLK_DIR */
+#define WM8993_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */
+#define WM8993_AIF_BCLK_INV_MASK 0x0100 /* AIF_BCLK_INV */
+#define WM8993_AIF_BCLK_INV_SHIFT 8 /* AIF_BCLK_INV */
+#define WM8993_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */
+#define WM8993_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */
+#define WM8993_AIF_LRCLK_INV_MASK 0x0080 /* AIF_LRCLK_INV */
+#define WM8993_AIF_LRCLK_INV_SHIFT 7 /* AIF_LRCLK_INV */
+#define WM8993_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */
+#define WM8993_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */
+#define WM8993_AIF_WL_SHIFT 5 /* AIF_WL - [6:5] */
+#define WM8993_AIF_WL_WIDTH 2 /* AIF_WL - [6:5] */
+#define WM8993_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */
+#define WM8993_AIF_FMT_SHIFT 3 /* AIF_FMT - [4:3] */
+#define WM8993_AIF_FMT_WIDTH 2 /* AIF_FMT - [4:3] */
+
+/*
+ * R5 (0x05) - Audio Interface (2)
+ */
+#define WM8993_AIFDACL_SRC 0x8000 /* AIFDACL_SRC */
+#define WM8993_AIFDACL_SRC_MASK 0x8000 /* AIFDACL_SRC */
+#define WM8993_AIFDACL_SRC_SHIFT 15 /* AIFDACL_SRC */
+#define WM8993_AIFDACL_SRC_WIDTH 1 /* AIFDACL_SRC */
+#define WM8993_AIFDACR_SRC 0x4000 /* AIFDACR_SRC */
+#define WM8993_AIFDACR_SRC_MASK 0x4000 /* AIFDACR_SRC */
+#define WM8993_AIFDACR_SRC_SHIFT 14 /* AIFDACR_SRC */
+#define WM8993_AIFDACR_SRC_WIDTH 1 /* AIFDACR_SRC */
+#define WM8993_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */
+#define WM8993_AIFDAC_TDM_MASK 0x2000 /* AIFDAC_TDM */
+#define WM8993_AIFDAC_TDM_SHIFT 13 /* AIFDAC_TDM */
+#define WM8993_AIFDAC_TDM_WIDTH 1 /* AIFDAC_TDM */
+#define WM8993_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8993_AIFDAC_TDM_CHAN_MASK 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8993_AIFDAC_TDM_CHAN_SHIFT 12 /* AIFDAC_TDM_CHAN */
+#define WM8993_AIFDAC_TDM_CHAN_WIDTH 1 /* AIFDAC_TDM_CHAN */
+#define WM8993_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST - [11:10] */
+#define WM8993_DAC_BOOST_SHIFT 10 /* DAC_BOOST - [11:10] */
+#define WM8993_DAC_BOOST_WIDTH 2 /* DAC_BOOST - [11:10] */
+#define WM8993_DAC_COMP 0x0010 /* DAC_COMP */
+#define WM8993_DAC_COMP_MASK 0x0010 /* DAC_COMP */
+#define WM8993_DAC_COMP_SHIFT 4 /* DAC_COMP */
+#define WM8993_DAC_COMP_WIDTH 1 /* DAC_COMP */
+#define WM8993_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */
+#define WM8993_DAC_COMPMODE_MASK 0x0008 /* DAC_COMPMODE */
+#define WM8993_DAC_COMPMODE_SHIFT 3 /* DAC_COMPMODE */
+#define WM8993_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */
+#define WM8993_ADC_COMP 0x0004 /* ADC_COMP */
+#define WM8993_ADC_COMP_MASK 0x0004 /* ADC_COMP */
+#define WM8993_ADC_COMP_SHIFT 2 /* ADC_COMP */
+#define WM8993_ADC_COMP_WIDTH 1 /* ADC_COMP */
+#define WM8993_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */
+#define WM8993_ADC_COMPMODE_MASK 0x0002 /* ADC_COMPMODE */
+#define WM8993_ADC_COMPMODE_SHIFT 1 /* ADC_COMPMODE */
+#define WM8993_ADC_COMPMODE_WIDTH 1 /* ADC_COMPMODE */
+#define WM8993_LOOPBACK 0x0001 /* LOOPBACK */
+#define WM8993_LOOPBACK_MASK 0x0001 /* LOOPBACK */
+#define WM8993_LOOPBACK_SHIFT 0 /* LOOPBACK */
+#define WM8993_LOOPBACK_WIDTH 1 /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clocking 1
+ */
+#define WM8993_TOCLK_RATE 0x8000 /* TOCLK_RATE */
+#define WM8993_TOCLK_RATE_MASK 0x8000 /* TOCLK_RATE */
+#define WM8993_TOCLK_RATE_SHIFT 15 /* TOCLK_RATE */
+#define WM8993_TOCLK_RATE_WIDTH 1 /* TOCLK_RATE */
+#define WM8993_TOCLK_ENA 0x4000 /* TOCLK_ENA */
+#define WM8993_TOCLK_ENA_MASK 0x4000 /* TOCLK_ENA */
+#define WM8993_TOCLK_ENA_SHIFT 14 /* TOCLK_ENA */
+#define WM8993_TOCLK_ENA_WIDTH 1 /* TOCLK_ENA */
+#define WM8993_OPCLK_DIV_MASK 0x1E00 /* OPCLK_DIV - [12:9] */
+#define WM8993_OPCLK_DIV_SHIFT 9 /* OPCLK_DIV - [12:9] */
+#define WM8993_OPCLK_DIV_WIDTH 4 /* OPCLK_DIV - [12:9] */
+#define WM8993_DCLK_DIV_MASK 0x01C0 /* DCLK_DIV - [8:6] */
+#define WM8993_DCLK_DIV_SHIFT 6 /* DCLK_DIV - [8:6] */
+#define WM8993_DCLK_DIV_WIDTH 3 /* DCLK_DIV - [8:6] */
+#define WM8993_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */
+#define WM8993_BCLK_DIV_SHIFT 1 /* BCLK_DIV - [4:1] */
+#define WM8993_BCLK_DIV_WIDTH 4 /* BCLK_DIV - [4:1] */
+
+/*
+ * R7 (0x07) - Clocking 2
+ */
+#define WM8993_MCLK_SRC 0x8000 /* MCLK_SRC */
+#define WM8993_MCLK_SRC_MASK 0x8000 /* MCLK_SRC */
+#define WM8993_MCLK_SRC_SHIFT 15 /* MCLK_SRC */
+#define WM8993_MCLK_SRC_WIDTH 1 /* MCLK_SRC */
+#define WM8993_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */
+#define WM8993_SYSCLK_SRC_MASK 0x4000 /* SYSCLK_SRC */
+#define WM8993_SYSCLK_SRC_SHIFT 14 /* SYSCLK_SRC */
+#define WM8993_SYSCLK_SRC_WIDTH 1 /* SYSCLK_SRC */
+#define WM8993_MCLK_DIV 0x1000 /* MCLK_DIV */
+#define WM8993_MCLK_DIV_MASK 0x1000 /* MCLK_DIV */
+#define WM8993_MCLK_DIV_SHIFT 12 /* MCLK_DIV */
+#define WM8993_MCLK_DIV_WIDTH 1 /* MCLK_DIV */
+#define WM8993_MCLK_INV 0x0400 /* MCLK_INV */
+#define WM8993_MCLK_INV_MASK 0x0400 /* MCLK_INV */
+#define WM8993_MCLK_INV_SHIFT 10 /* MCLK_INV */
+#define WM8993_MCLK_INV_WIDTH 1 /* MCLK_INV */
+#define WM8993_ADC_DIV_MASK 0x00E0 /* ADC_DIV - [7:5] */
+#define WM8993_ADC_DIV_SHIFT 5 /* ADC_DIV - [7:5] */
+#define WM8993_ADC_DIV_WIDTH 3 /* ADC_DIV - [7:5] */
+#define WM8993_DAC_DIV_MASK 0x001C /* DAC_DIV - [4:2] */
+#define WM8993_DAC_DIV_SHIFT 2 /* DAC_DIV - [4:2] */
+#define WM8993_DAC_DIV_WIDTH 3 /* DAC_DIV - [4:2] */
+
+/*
+ * R8 (0x08) - Audio Interface (3)
+ */
+#define WM8993_AIF_MSTR1 0x8000 /* AIF_MSTR1 */
+#define WM8993_AIF_MSTR1_MASK 0x8000 /* AIF_MSTR1 */
+#define WM8993_AIF_MSTR1_SHIFT 15 /* AIF_MSTR1 */
+#define WM8993_AIF_MSTR1_WIDTH 1 /* AIF_MSTR1 */
+
+/*
+ * R9 (0x09) - Audio Interface (4)
+ */
+#define WM8993_AIF_TRIS 0x2000 /* AIF_TRIS */
+#define WM8993_AIF_TRIS_MASK 0x2000 /* AIF_TRIS */
+#define WM8993_AIF_TRIS_SHIFT 13 /* AIF_TRIS */
+#define WM8993_AIF_TRIS_WIDTH 1 /* AIF_TRIS */
+#define WM8993_LRCLK_DIR 0x0800 /* LRCLK_DIR */
+#define WM8993_LRCLK_DIR_MASK 0x0800 /* LRCLK_DIR */
+#define WM8993_LRCLK_DIR_SHIFT 11 /* LRCLK_DIR */
+#define WM8993_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */
+#define WM8993_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */
+#define WM8993_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */
+#define WM8993_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */
+
+/*
+ * R10 (0x0A) - DAC CTRL
+ */
+#define WM8993_DAC_OSR128 0x2000 /* DAC_OSR128 */
+#define WM8993_DAC_OSR128_MASK 0x2000 /* DAC_OSR128 */
+#define WM8993_DAC_OSR128_SHIFT 13 /* DAC_OSR128 */
+#define WM8993_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */
+#define WM8993_DAC_MONO 0x0200 /* DAC_MONO */
+#define WM8993_DAC_MONO_MASK 0x0200 /* DAC_MONO */
+#define WM8993_DAC_MONO_SHIFT 9 /* DAC_MONO */
+#define WM8993_DAC_MONO_WIDTH 1 /* DAC_MONO */
+#define WM8993_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */
+#define WM8993_DAC_SB_FILT_MASK 0x0100 /* DAC_SB_FILT */
+#define WM8993_DAC_SB_FILT_SHIFT 8 /* DAC_SB_FILT */
+#define WM8993_DAC_SB_FILT_WIDTH 1 /* DAC_SB_FILT */
+#define WM8993_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */
+#define WM8993_DAC_MUTERATE_MASK 0x0080 /* DAC_MUTERATE */
+#define WM8993_DAC_MUTERATE_SHIFT 7 /* DAC_MUTERATE */
+#define WM8993_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */
+#define WM8993_DAC_UNMUTE_RAMP 0x0040 /* DAC_UNMUTE_RAMP */
+#define WM8993_DAC_UNMUTE_RAMP_MASK 0x0040 /* DAC_UNMUTE_RAMP */
+#define WM8993_DAC_UNMUTE_RAMP_SHIFT 6 /* DAC_UNMUTE_RAMP */
+#define WM8993_DAC_UNMUTE_RAMP_WIDTH 1 /* DAC_UNMUTE_RAMP */
+#define WM8993_DEEMPH_MASK 0x0030 /* DEEMPH - [5:4] */
+#define WM8993_DEEMPH_SHIFT 4 /* DEEMPH - [5:4] */
+#define WM8993_DEEMPH_WIDTH 2 /* DEEMPH - [5:4] */
+#define WM8993_DAC_MUTE 0x0004 /* DAC_MUTE */
+#define WM8993_DAC_MUTE_MASK 0x0004 /* DAC_MUTE */
+#define WM8993_DAC_MUTE_SHIFT 2 /* DAC_MUTE */
+#define WM8993_DAC_MUTE_WIDTH 1 /* DAC_MUTE */
+#define WM8993_DACL_DATINV 0x0002 /* DACL_DATINV */
+#define WM8993_DACL_DATINV_MASK 0x0002 /* DACL_DATINV */
+#define WM8993_DACL_DATINV_SHIFT 1 /* DACL_DATINV */
+#define WM8993_DACL_DATINV_WIDTH 1 /* DACL_DATINV */
+#define WM8993_DACR_DATINV 0x0001 /* DACR_DATINV */
+#define WM8993_DACR_DATINV_MASK 0x0001 /* DACR_DATINV */
+#define WM8993_DACR_DATINV_SHIFT 0 /* DACR_DATINV */
+#define WM8993_DACR_DATINV_WIDTH 1 /* DACR_DATINV */
+
+/*
+ * R11 (0x0B) - Left DAC Digital Volume
+ */
+#define WM8993_DAC_VU 0x0100 /* DAC_VU */
+#define WM8993_DAC_VU_MASK 0x0100 /* DAC_VU */
+#define WM8993_DAC_VU_SHIFT 8 /* DAC_VU */
+#define WM8993_DAC_VU_WIDTH 1 /* DAC_VU */
+#define WM8993_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */
+#define WM8993_DACL_VOL_SHIFT 0 /* DACL_VOL - [7:0] */
+#define WM8993_DACL_VOL_WIDTH 8 /* DACL_VOL - [7:0] */
+
+/*
+ * R12 (0x0C) - Right DAC Digital Volume
+ */
+#define WM8993_DAC_VU 0x0100 /* DAC_VU */
+#define WM8993_DAC_VU_MASK 0x0100 /* DAC_VU */
+#define WM8993_DAC_VU_SHIFT 8 /* DAC_VU */
+#define WM8993_DAC_VU_WIDTH 1 /* DAC_VU */
+#define WM8993_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */
+#define WM8993_DACR_VOL_SHIFT 0 /* DACR_VOL - [7:0] */
+#define WM8993_DACR_VOL_WIDTH 8 /* DACR_VOL - [7:0] */
+
+/*
+ * R13 (0x0D) - Digital Side Tone
+ */
+#define WM8993_ADCL_DAC_SVOL_MASK 0x1E00 /* ADCL_DAC_SVOL - [12:9] */
+#define WM8993_ADCL_DAC_SVOL_SHIFT 9 /* ADCL_DAC_SVOL - [12:9] */
+#define WM8993_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [12:9] */
+#define WM8993_ADCR_DAC_SVOL_MASK 0x01E0 /* ADCR_DAC_SVOL - [8:5] */
+#define WM8993_ADCR_DAC_SVOL_SHIFT 5 /* ADCR_DAC_SVOL - [8:5] */
+#define WM8993_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [8:5] */
+#define WM8993_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */
+#define WM8993_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */
+#define WM8993_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */
+#define WM8993_ADC_TO_DACR_MASK 0x0003 /* ADC_TO_DACR - [1:0] */
+#define WM8993_ADC_TO_DACR_SHIFT 0 /* ADC_TO_DACR - [1:0] */
+#define WM8993_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [1:0] */
+
+/*
+ * R14 (0x0E) - ADC CTRL
+ */
+#define WM8993_ADC_OSR128 0x0200 /* ADC_OSR128 */
+#define WM8993_ADC_OSR128_MASK 0x0200 /* ADC_OSR128 */
+#define WM8993_ADC_OSR128_SHIFT 9 /* ADC_OSR128 */
+#define WM8993_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */
+#define WM8993_ADC_HPF 0x0100 /* ADC_HPF */
+#define WM8993_ADC_HPF_MASK 0x0100 /* ADC_HPF */
+#define WM8993_ADC_HPF_SHIFT 8 /* ADC_HPF */
+#define WM8993_ADC_HPF_WIDTH 1 /* ADC_HPF */
+#define WM8993_ADC_HPF_CUT_MASK 0x0060 /* ADC_HPF_CUT - [6:5] */
+#define WM8993_ADC_HPF_CUT_SHIFT 5 /* ADC_HPF_CUT - [6:5] */
+#define WM8993_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [6:5] */
+#define WM8993_ADCL_DATINV 0x0002 /* ADCL_DATINV */
+#define WM8993_ADCL_DATINV_MASK 0x0002 /* ADCL_DATINV */
+#define WM8993_ADCL_DATINV_SHIFT 1 /* ADCL_DATINV */
+#define WM8993_ADCL_DATINV_WIDTH 1 /* ADCL_DATINV */
+#define WM8993_ADCR_DATINV 0x0001 /* ADCR_DATINV */
+#define WM8993_ADCR_DATINV_MASK 0x0001 /* ADCR_DATINV */
+#define WM8993_ADCR_DATINV_SHIFT 0 /* ADCR_DATINV */
+#define WM8993_ADCR_DATINV_WIDTH 1 /* ADCR_DATINV */
+
+/*
+ * R15 (0x0F) - Left ADC Digital Volume
+ */
+#define WM8993_ADC_VU 0x0100 /* ADC_VU */
+#define WM8993_ADC_VU_MASK 0x0100 /* ADC_VU */
+#define WM8993_ADC_VU_SHIFT 8 /* ADC_VU */
+#define WM8993_ADC_VU_WIDTH 1 /* ADC_VU */
+#define WM8993_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */
+#define WM8993_ADCL_VOL_SHIFT 0 /* ADCL_VOL - [7:0] */
+#define WM8993_ADCL_VOL_WIDTH 8 /* ADCL_VOL - [7:0] */
+
+/*
+ * R16 (0x10) - Right ADC Digital Volume
+ */
+#define WM8993_ADC_VU 0x0100 /* ADC_VU */
+#define WM8993_ADC_VU_MASK 0x0100 /* ADC_VU */
+#define WM8993_ADC_VU_SHIFT 8 /* ADC_VU */
+#define WM8993_ADC_VU_WIDTH 1 /* ADC_VU */
+#define WM8993_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */
+#define WM8993_ADCR_VOL_SHIFT 0 /* ADCR_VOL - [7:0] */
+#define WM8993_ADCR_VOL_WIDTH 8 /* ADCR_VOL - [7:0] */
+
+/*
+ * R18 (0x12) - GPIO CTRL 1
+ */
+#define WM8993_JD2_SC_EINT 0x8000 /* JD2_SC_EINT */
+#define WM8993_JD2_SC_EINT_MASK 0x8000 /* JD2_SC_EINT */
+#define WM8993_JD2_SC_EINT_SHIFT 15 /* JD2_SC_EINT */
+#define WM8993_JD2_SC_EINT_WIDTH 1 /* JD2_SC_EINT */
+#define WM8993_JD2_EINT 0x4000 /* JD2_EINT */
+#define WM8993_JD2_EINT_MASK 0x4000 /* JD2_EINT */
+#define WM8993_JD2_EINT_SHIFT 14 /* JD2_EINT */
+#define WM8993_JD2_EINT_WIDTH 1 /* JD2_EINT */
+#define WM8993_WSEQ_EINT 0x2000 /* WSEQ_EINT */
+#define WM8993_WSEQ_EINT_MASK 0x2000 /* WSEQ_EINT */
+#define WM8993_WSEQ_EINT_SHIFT 13 /* WSEQ_EINT */
+#define WM8993_WSEQ_EINT_WIDTH 1 /* WSEQ_EINT */
+#define WM8993_IRQ 0x1000 /* IRQ */
+#define WM8993_IRQ_MASK 0x1000 /* IRQ */
+#define WM8993_IRQ_SHIFT 12 /* IRQ */
+#define WM8993_IRQ_WIDTH 1 /* IRQ */
+#define WM8993_TEMPOK_EINT 0x0800 /* TEMPOK_EINT */
+#define WM8993_TEMPOK_EINT_MASK 0x0800 /* TEMPOK_EINT */
+#define WM8993_TEMPOK_EINT_SHIFT 11 /* TEMPOK_EINT */
+#define WM8993_TEMPOK_EINT_WIDTH 1 /* TEMPOK_EINT */
+#define WM8993_JD1_SC_EINT 0x0400 /* JD1_SC_EINT */
+#define WM8993_JD1_SC_EINT_MASK 0x0400 /* JD1_SC_EINT */
+#define WM8993_JD1_SC_EINT_SHIFT 10 /* JD1_SC_EINT */
+#define WM8993_JD1_SC_EINT_WIDTH 1 /* JD1_SC_EINT */
+#define WM8993_JD1_EINT 0x0200 /* JD1_EINT */
+#define WM8993_JD1_EINT_MASK 0x0200 /* JD1_EINT */
+#define WM8993_JD1_EINT_SHIFT 9 /* JD1_EINT */
+#define WM8993_JD1_EINT_WIDTH 1 /* JD1_EINT */
+#define WM8993_FLL_LOCK_EINT 0x0100 /* FLL_LOCK_EINT */
+#define WM8993_FLL_LOCK_EINT_MASK 0x0100 /* FLL_LOCK_EINT */
+#define WM8993_FLL_LOCK_EINT_SHIFT 8 /* FLL_LOCK_EINT */
+#define WM8993_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */
+#define WM8993_GPI8_EINT 0x0080 /* GPI8_EINT */
+#define WM8993_GPI8_EINT_MASK 0x0080 /* GPI8_EINT */
+#define WM8993_GPI8_EINT_SHIFT 7 /* GPI8_EINT */
+#define WM8993_GPI8_EINT_WIDTH 1 /* GPI8_EINT */
+#define WM8993_GPI7_EINT 0x0040 /* GPI7_EINT */
+#define WM8993_GPI7_EINT_MASK 0x0040 /* GPI7_EINT */
+#define WM8993_GPI7_EINT_SHIFT 6 /* GPI7_EINT */
+#define WM8993_GPI7_EINT_WIDTH 1 /* GPI7_EINT */
+#define WM8993_GPIO1_EINT 0x0001 /* GPIO1_EINT */
+#define WM8993_GPIO1_EINT_MASK 0x0001 /* GPIO1_EINT */
+#define WM8993_GPIO1_EINT_SHIFT 0 /* GPIO1_EINT */
+#define WM8993_GPIO1_EINT_WIDTH 1 /* GPIO1_EINT */
+
+/*
+ * R19 (0x13) - GPIO1
+ */
+#define WM8993_GPIO1_PU 0x0020 /* GPIO1_PU */
+#define WM8993_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */
+#define WM8993_GPIO1_PU_SHIFT 5 /* GPIO1_PU */
+#define WM8993_GPIO1_PU_WIDTH 1 /* GPIO1_PU */
+#define WM8993_GPIO1_PD 0x0010 /* GPIO1_PD */
+#define WM8993_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */
+#define WM8993_GPIO1_PD_SHIFT 4 /* GPIO1_PD */
+#define WM8993_GPIO1_PD_WIDTH 1 /* GPIO1_PD */
+#define WM8993_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */
+#define WM8993_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */
+#define WM8993_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */
+
+/*
+ * R20 (0x14) - IRQ_DEBOUNCE
+ */
+#define WM8993_JD2_SC_DB 0x8000 /* JD2_SC_DB */
+#define WM8993_JD2_SC_DB_MASK 0x8000 /* JD2_SC_DB */
+#define WM8993_JD2_SC_DB_SHIFT 15 /* JD2_SC_DB */
+#define WM8993_JD2_SC_DB_WIDTH 1 /* JD2_SC_DB */
+#define WM8993_JD2_DB 0x4000 /* JD2_DB */
+#define WM8993_JD2_DB_MASK 0x4000 /* JD2_DB */
+#define WM8993_JD2_DB_SHIFT 14 /* JD2_DB */
+#define WM8993_JD2_DB_WIDTH 1 /* JD2_DB */
+#define WM8993_WSEQ_DB 0x2000 /* WSEQ_DB */
+#define WM8993_WSEQ_DB_MASK 0x2000 /* WSEQ_DB */
+#define WM8993_WSEQ_DB_SHIFT 13 /* WSEQ_DB */
+#define WM8993_WSEQ_DB_WIDTH 1 /* WSEQ_DB */
+#define WM8993_TEMPOK_DB 0x0800 /* TEMPOK_DB */
+#define WM8993_TEMPOK_DB_MASK 0x0800 /* TEMPOK_DB */
+#define WM8993_TEMPOK_DB_SHIFT 11 /* TEMPOK_DB */
+#define WM8993_TEMPOK_DB_WIDTH 1 /* TEMPOK_DB */
+#define WM8993_JD1_SC_DB 0x0400 /* JD1_SC_DB */
+#define WM8993_JD1_SC_DB_MASK 0x0400 /* JD1_SC_DB */
+#define WM8993_JD1_SC_DB_SHIFT 10 /* JD1_SC_DB */
+#define WM8993_JD1_SC_DB_WIDTH 1 /* JD1_SC_DB */
+#define WM8993_JD1_DB 0x0200 /* JD1_DB */
+#define WM8993_JD1_DB_MASK 0x0200 /* JD1_DB */
+#define WM8993_JD1_DB_SHIFT 9 /* JD1_DB */
+#define WM8993_JD1_DB_WIDTH 1 /* JD1_DB */
+#define WM8993_FLL_LOCK_DB 0x0100 /* FLL_LOCK_DB */
+#define WM8993_FLL_LOCK_DB_MASK 0x0100 /* FLL_LOCK_DB */
+#define WM8993_FLL_LOCK_DB_SHIFT 8 /* FLL_LOCK_DB */
+#define WM8993_FLL_LOCK_DB_WIDTH 1 /* FLL_LOCK_DB */
+#define WM8993_GPI8_DB 0x0080 /* GPI8_DB */
+#define WM8993_GPI8_DB_MASK 0x0080 /* GPI8_DB */
+#define WM8993_GPI8_DB_SHIFT 7 /* GPI8_DB */
+#define WM8993_GPI8_DB_WIDTH 1 /* GPI8_DB */
+#define WM8993_GPI7_DB 0x0008 /* GPI7_DB */
+#define WM8993_GPI7_DB_MASK 0x0008 /* GPI7_DB */
+#define WM8993_GPI7_DB_SHIFT 3 /* GPI7_DB */
+#define WM8993_GPI7_DB_WIDTH 1 /* GPI7_DB */
+#define WM8993_GPIO1_DB 0x0001 /* GPIO1_DB */
+#define WM8993_GPIO1_DB_MASK 0x0001 /* GPIO1_DB */
+#define WM8993_GPIO1_DB_SHIFT 0 /* GPIO1_DB */
+#define WM8993_GPIO1_DB_WIDTH 1 /* GPIO1_DB */
+
+/*
+ * R22 (0x16) - GPIOCTRL 2
+ */
+#define WM8993_IM_JD2_EINT 0x2000 /* IM_JD2_EINT */
+#define WM8993_IM_JD2_EINT_MASK 0x2000 /* IM_JD2_EINT */
+#define WM8993_IM_JD2_EINT_SHIFT 13 /* IM_JD2_EINT */
+#define WM8993_IM_JD2_EINT_WIDTH 1 /* IM_JD2_EINT */
+#define WM8993_IM_JD2_SC_EINT 0x1000 /* IM_JD2_SC_EINT */
+#define WM8993_IM_JD2_SC_EINT_MASK 0x1000 /* IM_JD2_SC_EINT */
+#define WM8993_IM_JD2_SC_EINT_SHIFT 12 /* IM_JD2_SC_EINT */
+#define WM8993_IM_JD2_SC_EINT_WIDTH 1 /* IM_JD2_SC_EINT */
+#define WM8993_IM_TEMPOK_EINT 0x0800 /* IM_TEMPOK_EINT */
+#define WM8993_IM_TEMPOK_EINT_MASK 0x0800 /* IM_TEMPOK_EINT */
+#define WM8993_IM_TEMPOK_EINT_SHIFT 11 /* IM_TEMPOK_EINT */
+#define WM8993_IM_TEMPOK_EINT_WIDTH 1 /* IM_TEMPOK_EINT */
+#define WM8993_IM_JD1_SC_EINT 0x0400 /* IM_JD1_SC_EINT */
+#define WM8993_IM_JD1_SC_EINT_MASK 0x0400 /* IM_JD1_SC_EINT */
+#define WM8993_IM_JD1_SC_EINT_SHIFT 10 /* IM_JD1_SC_EINT */
+#define WM8993_IM_JD1_SC_EINT_WIDTH 1 /* IM_JD1_SC_EINT */
+#define WM8993_IM_JD1_EINT 0x0200 /* IM_JD1_EINT */
+#define WM8993_IM_JD1_EINT_MASK 0x0200 /* IM_JD1_EINT */
+#define WM8993_IM_JD1_EINT_SHIFT 9 /* IM_JD1_EINT */
+#define WM8993_IM_JD1_EINT_WIDTH 1 /* IM_JD1_EINT */
+#define WM8993_IM_FLL_LOCK_EINT 0x0100 /* IM_FLL_LOCK_EINT */
+#define WM8993_IM_FLL_LOCK_EINT_MASK 0x0100 /* IM_FLL_LOCK_EINT */
+#define WM8993_IM_FLL_LOCK_EINT_SHIFT 8 /* IM_FLL_LOCK_EINT */
+#define WM8993_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */
+#define WM8993_IM_GPI8_EINT 0x0040 /* IM_GPI8_EINT */
+#define WM8993_IM_GPI8_EINT_MASK 0x0040 /* IM_GPI8_EINT */
+#define WM8993_IM_GPI8_EINT_SHIFT 6 /* IM_GPI8_EINT */
+#define WM8993_IM_GPI8_EINT_WIDTH 1 /* IM_GPI8_EINT */
+#define WM8993_IM_GPIO1_EINT 0x0020 /* IM_GPIO1_EINT */
+#define WM8993_IM_GPIO1_EINT_MASK 0x0020 /* IM_GPIO1_EINT */
+#define WM8993_IM_GPIO1_EINT_SHIFT 5 /* IM_GPIO1_EINT */
+#define WM8993_IM_GPIO1_EINT_WIDTH 1 /* IM_GPIO1_EINT */
+#define WM8993_GPI8_ENA 0x0010 /* GPI8_ENA */
+#define WM8993_GPI8_ENA_MASK 0x0010 /* GPI8_ENA */
+#define WM8993_GPI8_ENA_SHIFT 4 /* GPI8_ENA */
+#define WM8993_GPI8_ENA_WIDTH 1 /* GPI8_ENA */
+#define WM8993_IM_GPI7_EINT 0x0004 /* IM_GPI7_EINT */
+#define WM8993_IM_GPI7_EINT_MASK 0x0004 /* IM_GPI7_EINT */
+#define WM8993_IM_GPI7_EINT_SHIFT 2 /* IM_GPI7_EINT */
+#define WM8993_IM_GPI7_EINT_WIDTH 1 /* IM_GPI7_EINT */
+#define WM8993_IM_WSEQ_EINT 0x0002 /* IM_WSEQ_EINT */
+#define WM8993_IM_WSEQ_EINT_MASK 0x0002 /* IM_WSEQ_EINT */
+#define WM8993_IM_WSEQ_EINT_SHIFT 1 /* IM_WSEQ_EINT */
+#define WM8993_IM_WSEQ_EINT_WIDTH 1 /* IM_WSEQ_EINT */
+#define WM8993_GPI7_ENA 0x0001 /* GPI7_ENA */
+#define WM8993_GPI7_ENA_MASK 0x0001 /* GPI7_ENA */
+#define WM8993_GPI7_ENA_SHIFT 0 /* GPI7_ENA */
+#define WM8993_GPI7_ENA_WIDTH 1 /* GPI7_ENA */
+
+/*
+ * R23 (0x17) - GPIO_POL
+ */
+#define WM8993_JD2_SC_POL 0x8000 /* JD2_SC_POL */
+#define WM8993_JD2_SC_POL_MASK 0x8000 /* JD2_SC_POL */
+#define WM8993_JD2_SC_POL_SHIFT 15 /* JD2_SC_POL */
+#define WM8993_JD2_SC_POL_WIDTH 1 /* JD2_SC_POL */
+#define WM8993_JD2_POL 0x4000 /* JD2_POL */
+#define WM8993_JD2_POL_MASK 0x4000 /* JD2_POL */
+#define WM8993_JD2_POL_SHIFT 14 /* JD2_POL */
+#define WM8993_JD2_POL_WIDTH 1 /* JD2_POL */
+#define WM8993_WSEQ_POL 0x2000 /* WSEQ_POL */
+#define WM8993_WSEQ_POL_MASK 0x2000 /* WSEQ_POL */
+#define WM8993_WSEQ_POL_SHIFT 13 /* WSEQ_POL */
+#define WM8993_WSEQ_POL_WIDTH 1 /* WSEQ_POL */
+#define WM8993_IRQ_POL 0x1000 /* IRQ_POL */
+#define WM8993_IRQ_POL_MASK 0x1000 /* IRQ_POL */
+#define WM8993_IRQ_POL_SHIFT 12 /* IRQ_POL */
+#define WM8993_IRQ_POL_WIDTH 1 /* IRQ_POL */
+#define WM8993_TEMPOK_POL 0x0800 /* TEMPOK_POL */
+#define WM8993_TEMPOK_POL_MASK 0x0800 /* TEMPOK_POL */
+#define WM8993_TEMPOK_POL_SHIFT 11 /* TEMPOK_POL */
+#define WM8993_TEMPOK_POL_WIDTH 1 /* TEMPOK_POL */
+#define WM8993_JD1_SC_POL 0x0400 /* JD1_SC_POL */
+#define WM8993_JD1_SC_POL_MASK 0x0400 /* JD1_SC_POL */
+#define WM8993_JD1_SC_POL_SHIFT 10 /* JD1_SC_POL */
+#define WM8993_JD1_SC_POL_WIDTH 1 /* JD1_SC_POL */
+#define WM8993_JD1_POL 0x0200 /* JD1_POL */
+#define WM8993_JD1_POL_MASK 0x0200 /* JD1_POL */
+#define WM8993_JD1_POL_SHIFT 9 /* JD1_POL */
+#define WM8993_JD1_POL_WIDTH 1 /* JD1_POL */
+#define WM8993_FLL_LOCK_POL 0x0100 /* FLL_LOCK_POL */
+#define WM8993_FLL_LOCK_POL_MASK 0x0100 /* FLL_LOCK_POL */
+#define WM8993_FLL_LOCK_POL_SHIFT 8 /* FLL_LOCK_POL */
+#define WM8993_FLL_LOCK_POL_WIDTH 1 /* FLL_LOCK_POL */
+#define WM8993_GPI8_POL 0x0080 /* GPI8_POL */
+#define WM8993_GPI8_POL_MASK 0x0080 /* GPI8_POL */
+#define WM8993_GPI8_POL_SHIFT 7 /* GPI8_POL */
+#define WM8993_GPI8_POL_WIDTH 1 /* GPI8_POL */
+#define WM8993_GPI7_POL 0x0040 /* GPI7_POL */
+#define WM8993_GPI7_POL_MASK 0x0040 /* GPI7_POL */
+#define WM8993_GPI7_POL_SHIFT 6 /* GPI7_POL */
+#define WM8993_GPI7_POL_WIDTH 1 /* GPI7_POL */
+#define WM8993_GPIO1_POL 0x0001 /* GPIO1_POL */
+#define WM8993_GPIO1_POL_MASK 0x0001 /* GPIO1_POL */
+#define WM8993_GPIO1_POL_SHIFT 0 /* GPIO1_POL */
+#define WM8993_GPIO1_POL_WIDTH 1 /* GPIO1_POL */
+
+/*
+ * R24 (0x18) - Left Line Input 1&2 Volume
+ */
+#define WM8993_IN1_VU 0x0100 /* IN1_VU */
+#define WM8993_IN1_VU_MASK 0x0100 /* IN1_VU */
+#define WM8993_IN1_VU_SHIFT 8 /* IN1_VU */
+#define WM8993_IN1_VU_WIDTH 1 /* IN1_VU */
+#define WM8993_IN1L_MUTE 0x0080 /* IN1L_MUTE */
+#define WM8993_IN1L_MUTE_MASK 0x0080 /* IN1L_MUTE */
+#define WM8993_IN1L_MUTE_SHIFT 7 /* IN1L_MUTE */
+#define WM8993_IN1L_MUTE_WIDTH 1 /* IN1L_MUTE */
+#define WM8993_IN1L_ZC 0x0040 /* IN1L_ZC */
+#define WM8993_IN1L_ZC_MASK 0x0040 /* IN1L_ZC */
+#define WM8993_IN1L_ZC_SHIFT 6 /* IN1L_ZC */
+#define WM8993_IN1L_ZC_WIDTH 1 /* IN1L_ZC */
+#define WM8993_IN1L_VOL_MASK 0x001F /* IN1L_VOL - [4:0] */
+#define WM8993_IN1L_VOL_SHIFT 0 /* IN1L_VOL - [4:0] */
+#define WM8993_IN1L_VOL_WIDTH 5 /* IN1L_VOL - [4:0] */
+
+/*
+ * R25 (0x19) - Left Line Input 3&4 Volume
+ */
+#define WM8993_IN2_VU 0x0100 /* IN2_VU */
+#define WM8993_IN2_VU_MASK 0x0100 /* IN2_VU */
+#define WM8993_IN2_VU_SHIFT 8 /* IN2_VU */
+#define WM8993_IN2_VU_WIDTH 1 /* IN2_VU */
+#define WM8993_IN2L_MUTE 0x0080 /* IN2L_MUTE */
+#define WM8993_IN2L_MUTE_MASK 0x0080 /* IN2L_MUTE */
+#define WM8993_IN2L_MUTE_SHIFT 7 /* IN2L_MUTE */
+#define WM8993_IN2L_MUTE_WIDTH 1 /* IN2L_MUTE */
+#define WM8993_IN2L_ZC 0x0040 /* IN2L_ZC */
+#define WM8993_IN2L_ZC_MASK 0x0040 /* IN2L_ZC */
+#define WM8993_IN2L_ZC_SHIFT 6 /* IN2L_ZC */
+#define WM8993_IN2L_ZC_WIDTH 1 /* IN2L_ZC */
+#define WM8993_IN2L_VOL_MASK 0x001F /* IN2L_VOL - [4:0] */
+#define WM8993_IN2L_VOL_SHIFT 0 /* IN2L_VOL - [4:0] */
+#define WM8993_IN2L_VOL_WIDTH 5 /* IN2L_VOL - [4:0] */
+
+/*
+ * R26 (0x1A) - Right Line Input 1&2 Volume
+ */
+#define WM8993_IN1_VU 0x0100 /* IN1_VU */
+#define WM8993_IN1_VU_MASK 0x0100 /* IN1_VU */
+#define WM8993_IN1_VU_SHIFT 8 /* IN1_VU */
+#define WM8993_IN1_VU_WIDTH 1 /* IN1_VU */
+#define WM8993_IN1R_MUTE 0x0080 /* IN1R_MUTE */
+#define WM8993_IN1R_MUTE_MASK 0x0080 /* IN1R_MUTE */
+#define WM8993_IN1R_MUTE_SHIFT 7 /* IN1R_MUTE */
+#define WM8993_IN1R_MUTE_WIDTH 1 /* IN1R_MUTE */
+#define WM8993_IN1R_ZC 0x0040 /* IN1R_ZC */
+#define WM8993_IN1R_ZC_MASK 0x0040 /* IN1R_ZC */
+#define WM8993_IN1R_ZC_SHIFT 6 /* IN1R_ZC */
+#define WM8993_IN1R_ZC_WIDTH 1 /* IN1R_ZC */
+#define WM8993_IN1R_VOL_MASK 0x001F /* IN1R_VOL - [4:0] */
+#define WM8993_IN1R_VOL_SHIFT 0 /* IN1R_VOL - [4:0] */
+#define WM8993_IN1R_VOL_WIDTH 5 /* IN1R_VOL - [4:0] */
+
+/*
+ * R27 (0x1B) - Right Line Input 3&4 Volume
+ */
+#define WM8993_IN2_VU 0x0100 /* IN2_VU */
+#define WM8993_IN2_VU_MASK 0x0100 /* IN2_VU */
+#define WM8993_IN2_VU_SHIFT 8 /* IN2_VU */
+#define WM8993_IN2_VU_WIDTH 1 /* IN2_VU */
+#define WM8993_IN2R_MUTE 0x0080 /* IN2R_MUTE */
+#define WM8993_IN2R_MUTE_MASK 0x0080 /* IN2R_MUTE */
+#define WM8993_IN2R_MUTE_SHIFT 7 /* IN2R_MUTE */
+#define WM8993_IN2R_MUTE_WIDTH 1 /* IN2R_MUTE */
+#define WM8993_IN2R_ZC 0x0040 /* IN2R_ZC */
+#define WM8993_IN2R_ZC_MASK 0x0040 /* IN2R_ZC */
+#define WM8993_IN2R_ZC_SHIFT 6 /* IN2R_ZC */
+#define WM8993_IN2R_ZC_WIDTH 1 /* IN2R_ZC */
+#define WM8993_IN2R_VOL_MASK 0x001F /* IN2R_VOL - [4:0] */
+#define WM8993_IN2R_VOL_SHIFT 0 /* IN2R_VOL - [4:0] */
+#define WM8993_IN2R_VOL_WIDTH 5 /* IN2R_VOL - [4:0] */
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM8993_HPOUT1_VU 0x0100 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_MASK 0x0100 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_SHIFT 8 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_WIDTH 1 /* HPOUT1_VU */
+#define WM8993_HPOUT1L_ZC 0x0080 /* HPOUT1L_ZC */
+#define WM8993_HPOUT1L_ZC_MASK 0x0080 /* HPOUT1L_ZC */
+#define WM8993_HPOUT1L_ZC_SHIFT 7 /* HPOUT1L_ZC */
+#define WM8993_HPOUT1L_ZC_WIDTH 1 /* HPOUT1L_ZC */
+#define WM8993_HPOUT1L_MUTE_N 0x0040 /* HPOUT1L_MUTE_N */
+#define WM8993_HPOUT1L_MUTE_N_MASK 0x0040 /* HPOUT1L_MUTE_N */
+#define WM8993_HPOUT1L_MUTE_N_SHIFT 6 /* HPOUT1L_MUTE_N */
+#define WM8993_HPOUT1L_MUTE_N_WIDTH 1 /* HPOUT1L_MUTE_N */
+#define WM8993_HPOUT1L_VOL_MASK 0x003F /* HPOUT1L_VOL - [5:0] */
+#define WM8993_HPOUT1L_VOL_SHIFT 0 /* HPOUT1L_VOL - [5:0] */
+#define WM8993_HPOUT1L_VOL_WIDTH 6 /* HPOUT1L_VOL - [5:0] */
+
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM8993_HPOUT1_VU 0x0100 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_MASK 0x0100 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_SHIFT 8 /* HPOUT1_VU */
+#define WM8993_HPOUT1_VU_WIDTH 1 /* HPOUT1_VU */
+#define WM8993_HPOUT1R_ZC 0x0080 /* HPOUT1R_ZC */
+#define WM8993_HPOUT1R_ZC_MASK 0x0080 /* HPOUT1R_ZC */
+#define WM8993_HPOUT1R_ZC_SHIFT 7 /* HPOUT1R_ZC */
+#define WM8993_HPOUT1R_ZC_WIDTH 1 /* HPOUT1R_ZC */
+#define WM8993_HPOUT1R_MUTE_N 0x0040 /* HPOUT1R_MUTE_N */
+#define WM8993_HPOUT1R_MUTE_N_MASK 0x0040 /* HPOUT1R_MUTE_N */
+#define WM8993_HPOUT1R_MUTE_N_SHIFT 6 /* HPOUT1R_MUTE_N */
+#define WM8993_HPOUT1R_MUTE_N_WIDTH 1 /* HPOUT1R_MUTE_N */
+#define WM8993_HPOUT1R_VOL_MASK 0x003F /* HPOUT1R_VOL - [5:0] */
+#define WM8993_HPOUT1R_VOL_SHIFT 0 /* HPOUT1R_VOL - [5:0] */
+#define WM8993_HPOUT1R_VOL_WIDTH 6 /* HPOUT1R_VOL - [5:0] */
+
+/*
+ * R30 (0x1E) - Line Outputs Volume
+ */
+#define WM8993_LINEOUT1N_MUTE 0x0040 /* LINEOUT1N_MUTE */
+#define WM8993_LINEOUT1N_MUTE_MASK 0x0040 /* LINEOUT1N_MUTE */
+#define WM8993_LINEOUT1N_MUTE_SHIFT 6 /* LINEOUT1N_MUTE */
+#define WM8993_LINEOUT1N_MUTE_WIDTH 1 /* LINEOUT1N_MUTE */
+#define WM8993_LINEOUT1P_MUTE 0x0020 /* LINEOUT1P_MUTE */
+#define WM8993_LINEOUT1P_MUTE_MASK 0x0020 /* LINEOUT1P_MUTE */
+#define WM8993_LINEOUT1P_MUTE_SHIFT 5 /* LINEOUT1P_MUTE */
+#define WM8993_LINEOUT1P_MUTE_WIDTH 1 /* LINEOUT1P_MUTE */
+#define WM8993_LINEOUT1_VOL 0x0010 /* LINEOUT1_VOL */
+#define WM8993_LINEOUT1_VOL_MASK 0x0010 /* LINEOUT1_VOL */
+#define WM8993_LINEOUT1_VOL_SHIFT 4 /* LINEOUT1_VOL */
+#define WM8993_LINEOUT1_VOL_WIDTH 1 /* LINEOUT1_VOL */
+#define WM8993_LINEOUT2N_MUTE 0x0004 /* LINEOUT2N_MUTE */
+#define WM8993_LINEOUT2N_MUTE_MASK 0x0004 /* LINEOUT2N_MUTE */
+#define WM8993_LINEOUT2N_MUTE_SHIFT 2 /* LINEOUT2N_MUTE */
+#define WM8993_LINEOUT2N_MUTE_WIDTH 1 /* LINEOUT2N_MUTE */
+#define WM8993_LINEOUT2P_MUTE 0x0002 /* LINEOUT2P_MUTE */
+#define WM8993_LINEOUT2P_MUTE_MASK 0x0002 /* LINEOUT2P_MUTE */
+#define WM8993_LINEOUT2P_MUTE_SHIFT 1 /* LINEOUT2P_MUTE */
+#define WM8993_LINEOUT2P_MUTE_WIDTH 1 /* LINEOUT2P_MUTE */
+#define WM8993_LINEOUT2_VOL 0x0001 /* LINEOUT2_VOL */
+#define WM8993_LINEOUT2_VOL_MASK 0x0001 /* LINEOUT2_VOL */
+#define WM8993_LINEOUT2_VOL_SHIFT 0 /* LINEOUT2_VOL */
+#define WM8993_LINEOUT2_VOL_WIDTH 1 /* LINEOUT2_VOL */
+
+/*
+ * R31 (0x1F) - HPOUT2 Volume
+ */
+#define WM8993_HPOUT2_MUTE 0x0020 /* HPOUT2_MUTE */
+#define WM8993_HPOUT2_MUTE_MASK 0x0020 /* HPOUT2_MUTE */
+#define WM8993_HPOUT2_MUTE_SHIFT 5 /* HPOUT2_MUTE */
+#define WM8993_HPOUT2_MUTE_WIDTH 1 /* HPOUT2_MUTE */
+#define WM8993_HPOUT2_VOL 0x0010 /* HPOUT2_VOL */
+#define WM8993_HPOUT2_VOL_MASK 0x0010 /* HPOUT2_VOL */
+#define WM8993_HPOUT2_VOL_SHIFT 4 /* HPOUT2_VOL */
+#define WM8993_HPOUT2_VOL_WIDTH 1 /* HPOUT2_VOL */
+
+/*
+ * R32 (0x20) - Left OPGA Volume
+ */
+#define WM8993_MIXOUT_VU 0x0100 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_MASK 0x0100 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_SHIFT 8 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_WIDTH 1 /* MIXOUT_VU */
+#define WM8993_MIXOUTL_ZC 0x0080 /* MIXOUTL_ZC */
+#define WM8993_MIXOUTL_ZC_MASK 0x0080 /* MIXOUTL_ZC */
+#define WM8993_MIXOUTL_ZC_SHIFT 7 /* MIXOUTL_ZC */
+#define WM8993_MIXOUTL_ZC_WIDTH 1 /* MIXOUTL_ZC */
+#define WM8993_MIXOUTL_MUTE_N 0x0040 /* MIXOUTL_MUTE_N */
+#define WM8993_MIXOUTL_MUTE_N_MASK 0x0040 /* MIXOUTL_MUTE_N */
+#define WM8993_MIXOUTL_MUTE_N_SHIFT 6 /* MIXOUTL_MUTE_N */
+#define WM8993_MIXOUTL_MUTE_N_WIDTH 1 /* MIXOUTL_MUTE_N */
+#define WM8993_MIXOUTL_VOL_MASK 0x003F /* MIXOUTL_VOL - [5:0] */
+#define WM8993_MIXOUTL_VOL_SHIFT 0 /* MIXOUTL_VOL - [5:0] */
+#define WM8993_MIXOUTL_VOL_WIDTH 6 /* MIXOUTL_VOL - [5:0] */
+
+/*
+ * R33 (0x21) - Right OPGA Volume
+ */
+#define WM8993_MIXOUT_VU 0x0100 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_MASK 0x0100 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_SHIFT 8 /* MIXOUT_VU */
+#define WM8993_MIXOUT_VU_WIDTH 1 /* MIXOUT_VU */
+#define WM8993_MIXOUTR_ZC 0x0080 /* MIXOUTR_ZC */
+#define WM8993_MIXOUTR_ZC_MASK 0x0080 /* MIXOUTR_ZC */
+#define WM8993_MIXOUTR_ZC_SHIFT 7 /* MIXOUTR_ZC */
+#define WM8993_MIXOUTR_ZC_WIDTH 1 /* MIXOUTR_ZC */
+#define WM8993_MIXOUTR_MUTE_N 0x0040 /* MIXOUTR_MUTE_N */
+#define WM8993_MIXOUTR_MUTE_N_MASK 0x0040 /* MIXOUTR_MUTE_N */
+#define WM8993_MIXOUTR_MUTE_N_SHIFT 6 /* MIXOUTR_MUTE_N */
+#define WM8993_MIXOUTR_MUTE_N_WIDTH 1 /* MIXOUTR_MUTE_N */
+#define WM8993_MIXOUTR_VOL_MASK 0x003F /* MIXOUTR_VOL - [5:0] */
+#define WM8993_MIXOUTR_VOL_SHIFT 0 /* MIXOUTR_VOL - [5:0] */
+#define WM8993_MIXOUTR_VOL_WIDTH 6 /* MIXOUTR_VOL - [5:0] */
+
+/*
+ * R34 (0x22) - SPKMIXL Attenuation
+ */
+#define WM8993_MIXINL_SPKMIXL_VOL 0x0020 /* MIXINL_SPKMIXL_VOL */
+#define WM8993_MIXINL_SPKMIXL_VOL_MASK 0x0020 /* MIXINL_SPKMIXL_VOL */
+#define WM8993_MIXINL_SPKMIXL_VOL_SHIFT 5 /* MIXINL_SPKMIXL_VOL */
+#define WM8993_MIXINL_SPKMIXL_VOL_WIDTH 1 /* MIXINL_SPKMIXL_VOL */
+#define WM8993_IN1LP_SPKMIXL_VOL 0x0010 /* IN1LP_SPKMIXL_VOL */
+#define WM8993_IN1LP_SPKMIXL_VOL_MASK 0x0010 /* IN1LP_SPKMIXL_VOL */
+#define WM8993_IN1LP_SPKMIXL_VOL_SHIFT 4 /* IN1LP_SPKMIXL_VOL */
+#define WM8993_IN1LP_SPKMIXL_VOL_WIDTH 1 /* IN1LP_SPKMIXL_VOL */
+#define WM8993_MIXOUTL_SPKMIXL_VOL 0x0008 /* MIXOUTL_SPKMIXL_VOL */
+#define WM8993_MIXOUTL_SPKMIXL_VOL_MASK 0x0008 /* MIXOUTL_SPKMIXL_VOL */
+#define WM8993_MIXOUTL_SPKMIXL_VOL_SHIFT 3 /* MIXOUTL_SPKMIXL_VOL */
+#define WM8993_MIXOUTL_SPKMIXL_VOL_WIDTH 1 /* MIXOUTL_SPKMIXL_VOL */
+#define WM8993_DACL_SPKMIXL_VOL 0x0004 /* DACL_SPKMIXL_VOL */
+#define WM8993_DACL_SPKMIXL_VOL_MASK 0x0004 /* DACL_SPKMIXL_VOL */
+#define WM8993_DACL_SPKMIXL_VOL_SHIFT 2 /* DACL_SPKMIXL_VOL */
+#define WM8993_DACL_SPKMIXL_VOL_WIDTH 1 /* DACL_SPKMIXL_VOL */
+#define WM8993_SPKMIXL_VOL_MASK 0x0003 /* SPKMIXL_VOL - [1:0] */
+#define WM8993_SPKMIXL_VOL_SHIFT 0 /* SPKMIXL_VOL - [1:0] */
+#define WM8993_SPKMIXL_VOL_WIDTH 2 /* SPKMIXL_VOL - [1:0] */
+
+/*
+ * R35 (0x23) - SPKMIXR Attenuation
+ */
+#define WM8993_SPKOUT_CLASSAB_MODE 0x0100 /* SPKOUT_CLASSAB_MODE */
+#define WM8993_SPKOUT_CLASSAB_MODE_MASK 0x0100 /* SPKOUT_CLASSAB_MODE */
+#define WM8993_SPKOUT_CLASSAB_MODE_SHIFT 8 /* SPKOUT_CLASSAB_MODE */
+#define WM8993_SPKOUT_CLASSAB_MODE_WIDTH 1 /* SPKOUT_CLASSAB_MODE */
+#define WM8993_MIXINR_SPKMIXR_VOL 0x0020 /* MIXINR_SPKMIXR_VOL */
+#define WM8993_MIXINR_SPKMIXR_VOL_MASK 0x0020 /* MIXINR_SPKMIXR_VOL */
+#define WM8993_MIXINR_SPKMIXR_VOL_SHIFT 5 /* MIXINR_SPKMIXR_VOL */
+#define WM8993_MIXINR_SPKMIXR_VOL_WIDTH 1 /* MIXINR_SPKMIXR_VOL */
+#define WM8993_IN1RP_SPKMIXR_VOL 0x0010 /* IN1RP_SPKMIXR_VOL */
+#define WM8993_IN1RP_SPKMIXR_VOL_MASK 0x0010 /* IN1RP_SPKMIXR_VOL */
+#define WM8993_IN1RP_SPKMIXR_VOL_SHIFT 4 /* IN1RP_SPKMIXR_VOL */
+#define WM8993_IN1RP_SPKMIXR_VOL_WIDTH 1 /* IN1RP_SPKMIXR_VOL */
+#define WM8993_MIXOUTR_SPKMIXR_VOL 0x0008 /* MIXOUTR_SPKMIXR_VOL */
+#define WM8993_MIXOUTR_SPKMIXR_VOL_MASK 0x0008 /* MIXOUTR_SPKMIXR_VOL */
+#define WM8993_MIXOUTR_SPKMIXR_VOL_SHIFT 3 /* MIXOUTR_SPKMIXR_VOL */
+#define WM8993_MIXOUTR_SPKMIXR_VOL_WIDTH 1 /* MIXOUTR_SPKMIXR_VOL */
+#define WM8993_DACR_SPKMIXR_VOL 0x0004 /* DACR_SPKMIXR_VOL */
+#define WM8993_DACR_SPKMIXR_VOL_MASK 0x0004 /* DACR_SPKMIXR_VOL */
+#define WM8993_DACR_SPKMIXR_VOL_SHIFT 2 /* DACR_SPKMIXR_VOL */
+#define WM8993_DACR_SPKMIXR_VOL_WIDTH 1 /* DACR_SPKMIXR_VOL */
+#define WM8993_SPKMIXR_VOL_MASK 0x0003 /* SPKMIXR_VOL - [1:0] */
+#define WM8993_SPKMIXR_VOL_SHIFT 0 /* SPKMIXR_VOL - [1:0] */
+#define WM8993_SPKMIXR_VOL_WIDTH 2 /* SPKMIXR_VOL - [1:0] */
+
+/*
+ * R36 (0x24) - SPKOUT Mixers
+ */
+#define WM8993_VRX_TO_SPKOUTL 0x0020 /* VRX_TO_SPKOUTL */
+#define WM8993_VRX_TO_SPKOUTL_MASK 0x0020 /* VRX_TO_SPKOUTL */
+#define WM8993_VRX_TO_SPKOUTL_SHIFT 5 /* VRX_TO_SPKOUTL */
+#define WM8993_VRX_TO_SPKOUTL_WIDTH 1 /* VRX_TO_SPKOUTL */
+#define WM8993_SPKMIXL_TO_SPKOUTL 0x0010 /* SPKMIXL_TO_SPKOUTL */
+#define WM8993_SPKMIXL_TO_SPKOUTL_MASK 0x0010 /* SPKMIXL_TO_SPKOUTL */
+#define WM8993_SPKMIXL_TO_SPKOUTL_SHIFT 4 /* SPKMIXL_TO_SPKOUTL */
+#define WM8993_SPKMIXL_TO_SPKOUTL_WIDTH 1 /* SPKMIXL_TO_SPKOUTL */
+#define WM8993_SPKMIXR_TO_SPKOUTL 0x0008 /* SPKMIXR_TO_SPKOUTL */
+#define WM8993_SPKMIXR_TO_SPKOUTL_MASK 0x0008 /* SPKMIXR_TO_SPKOUTL */
+#define WM8993_SPKMIXR_TO_SPKOUTL_SHIFT 3 /* SPKMIXR_TO_SPKOUTL */
+#define WM8993_SPKMIXR_TO_SPKOUTL_WIDTH 1 /* SPKMIXR_TO_SPKOUTL */
+#define WM8993_VRX_TO_SPKOUTR 0x0004 /* VRX_TO_SPKOUTR */
+#define WM8993_VRX_TO_SPKOUTR_MASK 0x0004 /* VRX_TO_SPKOUTR */
+#define WM8993_VRX_TO_SPKOUTR_SHIFT 2 /* VRX_TO_SPKOUTR */
+#define WM8993_VRX_TO_SPKOUTR_WIDTH 1 /* VRX_TO_SPKOUTR */
+#define WM8993_SPKMIXL_TO_SPKOUTR 0x0002 /* SPKMIXL_TO_SPKOUTR */
+#define WM8993_SPKMIXL_TO_SPKOUTR_MASK 0x0002 /* SPKMIXL_TO_SPKOUTR */
+#define WM8993_SPKMIXL_TO_SPKOUTR_SHIFT 1 /* SPKMIXL_TO_SPKOUTR */
+#define WM8993_SPKMIXL_TO_SPKOUTR_WIDTH 1 /* SPKMIXL_TO_SPKOUTR */
+#define WM8993_SPKMIXR_TO_SPKOUTR 0x0001 /* SPKMIXR_TO_SPKOUTR */
+#define WM8993_SPKMIXR_TO_SPKOUTR_MASK 0x0001 /* SPKMIXR_TO_SPKOUTR */
+#define WM8993_SPKMIXR_TO_SPKOUTR_SHIFT 0 /* SPKMIXR_TO_SPKOUTR */
+#define WM8993_SPKMIXR_TO_SPKOUTR_WIDTH 1 /* SPKMIXR_TO_SPKOUTR */
+
+/*
+ * R37 (0x25) - SPKOUT Boost
+ */
+#define WM8993_SPKOUTL_BOOST_MASK 0x0038 /* SPKOUTL_BOOST - [5:3] */
+#define WM8993_SPKOUTL_BOOST_SHIFT 3 /* SPKOUTL_BOOST - [5:3] */
+#define WM8993_SPKOUTL_BOOST_WIDTH 3 /* SPKOUTL_BOOST - [5:3] */
+#define WM8993_SPKOUTR_BOOST_MASK 0x0007 /* SPKOUTR_BOOST - [2:0] */
+#define WM8993_SPKOUTR_BOOST_SHIFT 0 /* SPKOUTR_BOOST - [2:0] */
+#define WM8993_SPKOUTR_BOOST_WIDTH 3 /* SPKOUTR_BOOST - [2:0] */
+
+/*
+ * R38 (0x26) - Speaker Volume Left
+ */
+#define WM8993_SPKOUT_VU 0x0100 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_MASK 0x0100 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_SHIFT 8 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_WIDTH 1 /* SPKOUT_VU */
+#define WM8993_SPKOUTL_ZC 0x0080 /* SPKOUTL_ZC */
+#define WM8993_SPKOUTL_ZC_MASK 0x0080 /* SPKOUTL_ZC */
+#define WM8993_SPKOUTL_ZC_SHIFT 7 /* SPKOUTL_ZC */
+#define WM8993_SPKOUTL_ZC_WIDTH 1 /* SPKOUTL_ZC */
+#define WM8993_SPKOUTL_MUTE_N 0x0040 /* SPKOUTL_MUTE_N */
+#define WM8993_SPKOUTL_MUTE_N_MASK 0x0040 /* SPKOUTL_MUTE_N */
+#define WM8993_SPKOUTL_MUTE_N_SHIFT 6 /* SPKOUTL_MUTE_N */
+#define WM8993_SPKOUTL_MUTE_N_WIDTH 1 /* SPKOUTL_MUTE_N */
+#define WM8993_SPKOUTL_VOL_MASK 0x003F /* SPKOUTL_VOL - [5:0] */
+#define WM8993_SPKOUTL_VOL_SHIFT 0 /* SPKOUTL_VOL - [5:0] */
+#define WM8993_SPKOUTL_VOL_WIDTH 6 /* SPKOUTL_VOL - [5:0] */
+
+/*
+ * R39 (0x27) - Speaker Volume Right
+ */
+#define WM8993_SPKOUT_VU 0x0100 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_MASK 0x0100 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_SHIFT 8 /* SPKOUT_VU */
+#define WM8993_SPKOUT_VU_WIDTH 1 /* SPKOUT_VU */
+#define WM8993_SPKOUTR_ZC 0x0080 /* SPKOUTR_ZC */
+#define WM8993_SPKOUTR_ZC_MASK 0x0080 /* SPKOUTR_ZC */
+#define WM8993_SPKOUTR_ZC_SHIFT 7 /* SPKOUTR_ZC */
+#define WM8993_SPKOUTR_ZC_WIDTH 1 /* SPKOUTR_ZC */
+#define WM8993_SPKOUTR_MUTE_N 0x0040 /* SPKOUTR_MUTE_N */
+#define WM8993_SPKOUTR_MUTE_N_MASK 0x0040 /* SPKOUTR_MUTE_N */
+#define WM8993_SPKOUTR_MUTE_N_SHIFT 6 /* SPKOUTR_MUTE_N */
+#define WM8993_SPKOUTR_MUTE_N_WIDTH 1 /* SPKOUTR_MUTE_N */
+#define WM8993_SPKOUTR_VOL_MASK 0x003F /* SPKOUTR_VOL - [5:0] */
+#define WM8993_SPKOUTR_VOL_SHIFT 0 /* SPKOUTR_VOL - [5:0] */
+#define WM8993_SPKOUTR_VOL_WIDTH 6 /* SPKOUTR_VOL - [5:0] */
+
+/*
+ * R40 (0x28) - Input Mixer2
+ */
+#define WM8993_IN2LP_TO_IN2L 0x0080 /* IN2LP_TO_IN2L */
+#define WM8993_IN2LP_TO_IN2L_MASK 0x0080 /* IN2LP_TO_IN2L */
+#define WM8993_IN2LP_TO_IN2L_SHIFT 7 /* IN2LP_TO_IN2L */
+#define WM8993_IN2LP_TO_IN2L_WIDTH 1 /* IN2LP_TO_IN2L */
+#define WM8993_IN2LN_TO_IN2L 0x0040 /* IN2LN_TO_IN2L */
+#define WM8993_IN2LN_TO_IN2L_MASK 0x0040 /* IN2LN_TO_IN2L */
+#define WM8993_IN2LN_TO_IN2L_SHIFT 6 /* IN2LN_TO_IN2L */
+#define WM8993_IN2LN_TO_IN2L_WIDTH 1 /* IN2LN_TO_IN2L */
+#define WM8993_IN1LP_TO_IN1L 0x0020 /* IN1LP_TO_IN1L */
+#define WM8993_IN1LP_TO_IN1L_MASK 0x0020 /* IN1LP_TO_IN1L */
+#define WM8993_IN1LP_TO_IN1L_SHIFT 5 /* IN1LP_TO_IN1L */
+#define WM8993_IN1LP_TO_IN1L_WIDTH 1 /* IN1LP_TO_IN1L */
+#define WM8993_IN1LN_TO_IN1L 0x0010 /* IN1LN_TO_IN1L */
+#define WM8993_IN1LN_TO_IN1L_MASK 0x0010 /* IN1LN_TO_IN1L */
+#define WM8993_IN1LN_TO_IN1L_SHIFT 4 /* IN1LN_TO_IN1L */
+#define WM8993_IN1LN_TO_IN1L_WIDTH 1 /* IN1LN_TO_IN1L */
+#define WM8993_IN2RP_TO_IN2R 0x0008 /* IN2RP_TO_IN2R */
+#define WM8993_IN2RP_TO_IN2R_MASK 0x0008 /* IN2RP_TO_IN2R */
+#define WM8993_IN2RP_TO_IN2R_SHIFT 3 /* IN2RP_TO_IN2R */
+#define WM8993_IN2RP_TO_IN2R_WIDTH 1 /* IN2RP_TO_IN2R */
+#define WM8993_IN2RN_TO_IN2R 0x0004 /* IN2RN_TO_IN2R */
+#define WM8993_IN2RN_TO_IN2R_MASK 0x0004 /* IN2RN_TO_IN2R */
+#define WM8993_IN2RN_TO_IN2R_SHIFT 2 /* IN2RN_TO_IN2R */
+#define WM8993_IN2RN_TO_IN2R_WIDTH 1 /* IN2RN_TO_IN2R */
+#define WM8993_IN1RP_TO_IN1R 0x0002 /* IN1RP_TO_IN1R */
+#define WM8993_IN1RP_TO_IN1R_MASK 0x0002 /* IN1RP_TO_IN1R */
+#define WM8993_IN1RP_TO_IN1R_SHIFT 1 /* IN1RP_TO_IN1R */
+#define WM8993_IN1RP_TO_IN1R_WIDTH 1 /* IN1RP_TO_IN1R */
+#define WM8993_IN1RN_TO_IN1R 0x0001 /* IN1RN_TO_IN1R */
+#define WM8993_IN1RN_TO_IN1R_MASK 0x0001 /* IN1RN_TO_IN1R */
+#define WM8993_IN1RN_TO_IN1R_SHIFT 0 /* IN1RN_TO_IN1R */
+#define WM8993_IN1RN_TO_IN1R_WIDTH 1 /* IN1RN_TO_IN1R */
+
+/*
+ * R41 (0x29) - Input Mixer3
+ */
+#define WM8993_IN2L_TO_MIXINL 0x0100 /* IN2L_TO_MIXINL */
+#define WM8993_IN2L_TO_MIXINL_MASK 0x0100 /* IN2L_TO_MIXINL */
+#define WM8993_IN2L_TO_MIXINL_SHIFT 8 /* IN2L_TO_MIXINL */
+#define WM8993_IN2L_TO_MIXINL_WIDTH 1 /* IN2L_TO_MIXINL */
+#define WM8993_IN2L_MIXINL_VOL 0x0080 /* IN2L_MIXINL_VOL */
+#define WM8993_IN2L_MIXINL_VOL_MASK 0x0080 /* IN2L_MIXINL_VOL */
+#define WM8993_IN2L_MIXINL_VOL_SHIFT 7 /* IN2L_MIXINL_VOL */
+#define WM8993_IN2L_MIXINL_VOL_WIDTH 1 /* IN2L_MIXINL_VOL */
+#define WM8993_IN1L_TO_MIXINL 0x0020 /* IN1L_TO_MIXINL */
+#define WM8993_IN1L_TO_MIXINL_MASK 0x0020 /* IN1L_TO_MIXINL */
+#define WM8993_IN1L_TO_MIXINL_SHIFT 5 /* IN1L_TO_MIXINL */
+#define WM8993_IN1L_TO_MIXINL_WIDTH 1 /* IN1L_TO_MIXINL */
+#define WM8993_IN1L_MIXINL_VOL 0x0010 /* IN1L_MIXINL_VOL */
+#define WM8993_IN1L_MIXINL_VOL_MASK 0x0010 /* IN1L_MIXINL_VOL */
+#define WM8993_IN1L_MIXINL_VOL_SHIFT 4 /* IN1L_MIXINL_VOL */
+#define WM8993_IN1L_MIXINL_VOL_WIDTH 1 /* IN1L_MIXINL_VOL */
+#define WM8993_MIXOUTL_MIXINL_VOL_MASK 0x0007 /* MIXOUTL_MIXINL_VOL - [2:0] */
+#define WM8993_MIXOUTL_MIXINL_VOL_SHIFT 0 /* MIXOUTL_MIXINL_VOL - [2:0] */
+#define WM8993_MIXOUTL_MIXINL_VOL_WIDTH 3 /* MIXOUTL_MIXINL_VOL - [2:0] */
+
+/*
+ * R42 (0x2A) - Input Mixer4
+ */
+#define WM8993_IN2R_TO_MIXINR 0x0100 /* IN2R_TO_MIXINR */
+#define WM8993_IN2R_TO_MIXINR_MASK 0x0100 /* IN2R_TO_MIXINR */
+#define WM8993_IN2R_TO_MIXINR_SHIFT 8 /* IN2R_TO_MIXINR */
+#define WM8993_IN2R_TO_MIXINR_WIDTH 1 /* IN2R_TO_MIXINR */
+#define WM8993_IN2R_MIXINR_VOL 0x0080 /* IN2R_MIXINR_VOL */
+#define WM8993_IN2R_MIXINR_VOL_MASK 0x0080 /* IN2R_MIXINR_VOL */
+#define WM8993_IN2R_MIXINR_VOL_SHIFT 7 /* IN2R_MIXINR_VOL */
+#define WM8993_IN2R_MIXINR_VOL_WIDTH 1 /* IN2R_MIXINR_VOL */
+#define WM8993_IN1R_TO_MIXINR 0x0020 /* IN1R_TO_MIXINR */
+#define WM8993_IN1R_TO_MIXINR_MASK 0x0020 /* IN1R_TO_MIXINR */
+#define WM8993_IN1R_TO_MIXINR_SHIFT 5 /* IN1R_TO_MIXINR */
+#define WM8993_IN1R_TO_MIXINR_WIDTH 1 /* IN1R_TO_MIXINR */
+#define WM8993_IN1R_MIXINR_VOL 0x0010 /* IN1R_MIXINR_VOL */
+#define WM8993_IN1R_MIXINR_VOL_MASK 0x0010 /* IN1R_MIXINR_VOL */
+#define WM8993_IN1R_MIXINR_VOL_SHIFT 4 /* IN1R_MIXINR_VOL */
+#define WM8993_IN1R_MIXINR_VOL_WIDTH 1 /* IN1R_MIXINR_VOL */
+#define WM8993_MIXOUTR_MIXINR_VOL_MASK 0x0007 /* MIXOUTR_MIXINR_VOL - [2:0] */
+#define WM8993_MIXOUTR_MIXINR_VOL_SHIFT 0 /* MIXOUTR_MIXINR_VOL - [2:0] */
+#define WM8993_MIXOUTR_MIXINR_VOL_WIDTH 3 /* MIXOUTR_MIXINR_VOL - [2:0] */
+
+/*
+ * R43 (0x2B) - Input Mixer5
+ */
+#define WM8993_IN1LP_MIXINL_VOL_MASK 0x01C0 /* IN1LP_MIXINL_VOL - [8:6] */
+#define WM8993_IN1LP_MIXINL_VOL_SHIFT 6 /* IN1LP_MIXINL_VOL - [8:6] */
+#define WM8993_IN1LP_MIXINL_VOL_WIDTH 3 /* IN1LP_MIXINL_VOL - [8:6] */
+#define WM8993_VRX_MIXINL_VOL_MASK 0x0007 /* VRX_MIXINL_VOL - [2:0] */
+#define WM8993_VRX_MIXINL_VOL_SHIFT 0 /* VRX_MIXINL_VOL - [2:0] */
+#define WM8993_VRX_MIXINL_VOL_WIDTH 3 /* VRX_MIXINL_VOL - [2:0] */
+
+/*
+ * R44 (0x2C) - Input Mixer6
+ */
+#define WM8993_IN1RP_MIXINR_VOL_MASK 0x01C0 /* IN1RP_MIXINR_VOL - [8:6] */
+#define WM8993_IN1RP_MIXINR_VOL_SHIFT 6 /* IN1RP_MIXINR_VOL - [8:6] */
+#define WM8993_IN1RP_MIXINR_VOL_WIDTH 3 /* IN1RP_MIXINR_VOL - [8:6] */
+#define WM8993_VRX_MIXINR_VOL_MASK 0x0007 /* VRX_MIXINR_VOL - [2:0] */
+#define WM8993_VRX_MIXINR_VOL_SHIFT 0 /* VRX_MIXINR_VOL - [2:0] */
+#define WM8993_VRX_MIXINR_VOL_WIDTH 3 /* VRX_MIXINR_VOL - [2:0] */
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM8993_DACL_TO_HPOUT1L 0x0100 /* DACL_TO_HPOUT1L */
+#define WM8993_DACL_TO_HPOUT1L_MASK 0x0100 /* DACL_TO_HPOUT1L */
+#define WM8993_DACL_TO_HPOUT1L_SHIFT 8 /* DACL_TO_HPOUT1L */
+#define WM8993_DACL_TO_HPOUT1L_WIDTH 1 /* DACL_TO_HPOUT1L */
+#define WM8993_MIXINR_TO_MIXOUTL 0x0080 /* MIXINR_TO_MIXOUTL */
+#define WM8993_MIXINR_TO_MIXOUTL_MASK 0x0080 /* MIXINR_TO_MIXOUTL */
+#define WM8993_MIXINR_TO_MIXOUTL_SHIFT 7 /* MIXINR_TO_MIXOUTL */
+#define WM8993_MIXINR_TO_MIXOUTL_WIDTH 1 /* MIXINR_TO_MIXOUTL */
+#define WM8993_MIXINL_TO_MIXOUTL 0x0040 /* MIXINL_TO_MIXOUTL */
+#define WM8993_MIXINL_TO_MIXOUTL_MASK 0x0040 /* MIXINL_TO_MIXOUTL */
+#define WM8993_MIXINL_TO_MIXOUTL_SHIFT 6 /* MIXINL_TO_MIXOUTL */
+#define WM8993_MIXINL_TO_MIXOUTL_WIDTH 1 /* MIXINL_TO_MIXOUTL */
+#define WM8993_IN2RN_TO_MIXOUTL 0x0020 /* IN2RN_TO_MIXOUTL */
+#define WM8993_IN2RN_TO_MIXOUTL_MASK 0x0020 /* IN2RN_TO_MIXOUTL */
+#define WM8993_IN2RN_TO_MIXOUTL_SHIFT 5 /* IN2RN_TO_MIXOUTL */
+#define WM8993_IN2RN_TO_MIXOUTL_WIDTH 1 /* IN2RN_TO_MIXOUTL */
+#define WM8993_IN2LN_TO_MIXOUTL 0x0010 /* IN2LN_TO_MIXOUTL */
+#define WM8993_IN2LN_TO_MIXOUTL_MASK 0x0010 /* IN2LN_TO_MIXOUTL */
+#define WM8993_IN2LN_TO_MIXOUTL_SHIFT 4 /* IN2LN_TO_MIXOUTL */
+#define WM8993_IN2LN_TO_MIXOUTL_WIDTH 1 /* IN2LN_TO_MIXOUTL */
+#define WM8993_IN1R_TO_MIXOUTL 0x0008 /* IN1R_TO_MIXOUTL */
+#define WM8993_IN1R_TO_MIXOUTL_MASK 0x0008 /* IN1R_TO_MIXOUTL */
+#define WM8993_IN1R_TO_MIXOUTL_SHIFT 3 /* IN1R_TO_MIXOUTL */
+#define WM8993_IN1R_TO_MIXOUTL_WIDTH 1 /* IN1R_TO_MIXOUTL */
+#define WM8993_IN1L_TO_MIXOUTL 0x0004 /* IN1L_TO_MIXOUTL */
+#define WM8993_IN1L_TO_MIXOUTL_MASK 0x0004 /* IN1L_TO_MIXOUTL */
+#define WM8993_IN1L_TO_MIXOUTL_SHIFT 2 /* IN1L_TO_MIXOUTL */
+#define WM8993_IN1L_TO_MIXOUTL_WIDTH 1 /* IN1L_TO_MIXOUTL */
+#define WM8993_IN2LP_TO_MIXOUTL 0x0002 /* IN2LP_TO_MIXOUTL */
+#define WM8993_IN2LP_TO_MIXOUTL_MASK 0x0002 /* IN2LP_TO_MIXOUTL */
+#define WM8993_IN2LP_TO_MIXOUTL_SHIFT 1 /* IN2LP_TO_MIXOUTL */
+#define WM8993_IN2LP_TO_MIXOUTL_WIDTH 1 /* IN2LP_TO_MIXOUTL */
+#define WM8993_DACL_TO_MIXOUTL 0x0001 /* DACL_TO_MIXOUTL */
+#define WM8993_DACL_TO_MIXOUTL_MASK 0x0001 /* DACL_TO_MIXOUTL */
+#define WM8993_DACL_TO_MIXOUTL_SHIFT 0 /* DACL_TO_MIXOUTL */
+#define WM8993_DACL_TO_MIXOUTL_WIDTH 1 /* DACL_TO_MIXOUTL */
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM8993_DACR_TO_HPOUT1R 0x0100 /* DACR_TO_HPOUT1R */
+#define WM8993_DACR_TO_HPOUT1R_MASK 0x0100 /* DACR_TO_HPOUT1R */
+#define WM8993_DACR_TO_HPOUT1R_SHIFT 8 /* DACR_TO_HPOUT1R */
+#define WM8993_DACR_TO_HPOUT1R_WIDTH 1 /* DACR_TO_HPOUT1R */
+#define WM8993_MIXINL_TO_MIXOUTR 0x0080 /* MIXINL_TO_MIXOUTR */
+#define WM8993_MIXINL_TO_MIXOUTR_MASK 0x0080 /* MIXINL_TO_MIXOUTR */
+#define WM8993_MIXINL_TO_MIXOUTR_SHIFT 7 /* MIXINL_TO_MIXOUTR */
+#define WM8993_MIXINL_TO_MIXOUTR_WIDTH 1 /* MIXINL_TO_MIXOUTR */
+#define WM8993_MIXINR_TO_MIXOUTR 0x0040 /* MIXINR_TO_MIXOUTR */
+#define WM8993_MIXINR_TO_MIXOUTR_MASK 0x0040 /* MIXINR_TO_MIXOUTR */
+#define WM8993_MIXINR_TO_MIXOUTR_SHIFT 6 /* MIXINR_TO_MIXOUTR */
+#define WM8993_MIXINR_TO_MIXOUTR_WIDTH 1 /* MIXINR_TO_MIXOUTR */
+#define WM8993_IN2LN_TO_MIXOUTR 0x0020 /* IN2LN_TO_MIXOUTR */
+#define WM8993_IN2LN_TO_MIXOUTR_MASK 0x0020 /* IN2LN_TO_MIXOUTR */
+#define WM8993_IN2LN_TO_MIXOUTR_SHIFT 5 /* IN2LN_TO_MIXOUTR */
+#define WM8993_IN2LN_TO_MIXOUTR_WIDTH 1 /* IN2LN_TO_MIXOUTR */
+#define WM8993_IN2RN_TO_MIXOUTR 0x0010 /* IN2RN_TO_MIXOUTR */
+#define WM8993_IN2RN_TO_MIXOUTR_MASK 0x0010 /* IN2RN_TO_MIXOUTR */
+#define WM8993_IN2RN_TO_MIXOUTR_SHIFT 4 /* IN2RN_TO_MIXOUTR */
+#define WM8993_IN2RN_TO_MIXOUTR_WIDTH 1 /* IN2RN_TO_MIXOUTR */
+#define WM8993_IN1L_TO_MIXOUTR 0x0008 /* IN1L_TO_MIXOUTR */
+#define WM8993_IN1L_TO_MIXOUTR_MASK 0x0008 /* IN1L_TO_MIXOUTR */
+#define WM8993_IN1L_TO_MIXOUTR_SHIFT 3 /* IN1L_TO_MIXOUTR */
+#define WM8993_IN1L_TO_MIXOUTR_WIDTH 1 /* IN1L_TO_MIXOUTR */
+#define WM8993_IN1R_TO_MIXOUTR 0x0004 /* IN1R_TO_MIXOUTR */
+#define WM8993_IN1R_TO_MIXOUTR_MASK 0x0004 /* IN1R_TO_MIXOUTR */
+#define WM8993_IN1R_TO_MIXOUTR_SHIFT 2 /* IN1R_TO_MIXOUTR */
+#define WM8993_IN1R_TO_MIXOUTR_WIDTH 1 /* IN1R_TO_MIXOUTR */
+#define WM8993_IN2RP_TO_MIXOUTR 0x0002 /* IN2RP_TO_MIXOUTR */
+#define WM8993_IN2RP_TO_MIXOUTR_MASK 0x0002 /* IN2RP_TO_MIXOUTR */
+#define WM8993_IN2RP_TO_MIXOUTR_SHIFT 1 /* IN2RP_TO_MIXOUTR */
+#define WM8993_IN2RP_TO_MIXOUTR_WIDTH 1 /* IN2RP_TO_MIXOUTR */
+#define WM8993_DACR_TO_MIXOUTR 0x0001 /* DACR_TO_MIXOUTR */
+#define WM8993_DACR_TO_MIXOUTR_MASK 0x0001 /* DACR_TO_MIXOUTR */
+#define WM8993_DACR_TO_MIXOUTR_SHIFT 0 /* DACR_TO_MIXOUTR */
+#define WM8993_DACR_TO_MIXOUTR_WIDTH 1 /* DACR_TO_MIXOUTR */
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM8993_IN2LP_MIXOUTL_VOL_MASK 0x0E00 /* IN2LP_MIXOUTL_VOL - [11:9] */
+#define WM8993_IN2LP_MIXOUTL_VOL_SHIFT 9 /* IN2LP_MIXOUTL_VOL - [11:9] */
+#define WM8993_IN2LP_MIXOUTL_VOL_WIDTH 3 /* IN2LP_MIXOUTL_VOL - [11:9] */
+#define WM8993_IN2LN_MIXOUTL_VOL_MASK 0x01C0 /* IN2LN_MIXOUTL_VOL - [8:6] */
+#define WM8993_IN2LN_MIXOUTL_VOL_SHIFT 6 /* IN2LN_MIXOUTL_VOL - [8:6] */
+#define WM8993_IN2LN_MIXOUTL_VOL_WIDTH 3 /* IN2LN_MIXOUTL_VOL - [8:6] */
+#define WM8993_IN1R_MIXOUTL_VOL_MASK 0x0038 /* IN1R_MIXOUTL_VOL - [5:3] */
+#define WM8993_IN1R_MIXOUTL_VOL_SHIFT 3 /* IN1R_MIXOUTL_VOL - [5:3] */
+#define WM8993_IN1R_MIXOUTL_VOL_WIDTH 3 /* IN1R_MIXOUTL_VOL - [5:3] */
+#define WM8993_IN1L_MIXOUTL_VOL_MASK 0x0007 /* IN1L_MIXOUTL_VOL - [2:0] */
+#define WM8993_IN1L_MIXOUTL_VOL_SHIFT 0 /* IN1L_MIXOUTL_VOL - [2:0] */
+#define WM8993_IN1L_MIXOUTL_VOL_WIDTH 3 /* IN1L_MIXOUTL_VOL - [2:0] */
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM8993_IN2RP_MIXOUTR_VOL_MASK 0x0E00 /* IN2RP_MIXOUTR_VOL - [11:9] */
+#define WM8993_IN2RP_MIXOUTR_VOL_SHIFT 9 /* IN2RP_MIXOUTR_VOL - [11:9] */
+#define WM8993_IN2RP_MIXOUTR_VOL_WIDTH 3 /* IN2RP_MIXOUTR_VOL - [11:9] */
+#define WM8993_IN2RN_MIXOUTR_VOL_MASK 0x01C0 /* IN2RN_MIXOUTR_VOL - [8:6] */
+#define WM8993_IN2RN_MIXOUTR_VOL_SHIFT 6 /* IN2RN_MIXOUTR_VOL - [8:6] */
+#define WM8993_IN2RN_MIXOUTR_VOL_WIDTH 3 /* IN2RN_MIXOUTR_VOL - [8:6] */
+#define WM8993_IN1L_MIXOUTR_VOL_MASK 0x0038 /* IN1L_MIXOUTR_VOL - [5:3] */
+#define WM8993_IN1L_MIXOUTR_VOL_SHIFT 3 /* IN1L_MIXOUTR_VOL - [5:3] */
+#define WM8993_IN1L_MIXOUTR_VOL_WIDTH 3 /* IN1L_MIXOUTR_VOL - [5:3] */
+#define WM8993_IN1R_MIXOUTR_VOL_MASK 0x0007 /* IN1R_MIXOUTR_VOL - [2:0] */
+#define WM8993_IN1R_MIXOUTR_VOL_SHIFT 0 /* IN1R_MIXOUTR_VOL - [2:0] */
+#define WM8993_IN1R_MIXOUTR_VOL_WIDTH 3 /* IN1R_MIXOUTR_VOL - [2:0] */
+
+/*
+ * R49 (0x31) - Output Mixer5
+ */
+#define WM8993_DACL_MIXOUTL_VOL_MASK 0x0E00 /* DACL_MIXOUTL_VOL - [11:9] */
+#define WM8993_DACL_MIXOUTL_VOL_SHIFT 9 /* DACL_MIXOUTL_VOL - [11:9] */
+#define WM8993_DACL_MIXOUTL_VOL_WIDTH 3 /* DACL_MIXOUTL_VOL - [11:9] */
+#define WM8993_IN2RN_MIXOUTL_VOL_MASK 0x01C0 /* IN2RN_MIXOUTL_VOL - [8:6] */
+#define WM8993_IN2RN_MIXOUTL_VOL_SHIFT 6 /* IN2RN_MIXOUTL_VOL - [8:6] */
+#define WM8993_IN2RN_MIXOUTL_VOL_WIDTH 3 /* IN2RN_MIXOUTL_VOL - [8:6] */
+#define WM8993_MIXINR_MIXOUTL_VOL_MASK 0x0038 /* MIXINR_MIXOUTL_VOL - [5:3] */
+#define WM8993_MIXINR_MIXOUTL_VOL_SHIFT 3 /* MIXINR_MIXOUTL_VOL - [5:3] */
+#define WM8993_MIXINR_MIXOUTL_VOL_WIDTH 3 /* MIXINR_MIXOUTL_VOL - [5:3] */
+#define WM8993_MIXINL_MIXOUTL_VOL_MASK 0x0007 /* MIXINL_MIXOUTL_VOL - [2:0] */
+#define WM8993_MIXINL_MIXOUTL_VOL_SHIFT 0 /* MIXINL_MIXOUTL_VOL - [2:0] */
+#define WM8993_MIXINL_MIXOUTL_VOL_WIDTH 3 /* MIXINL_MIXOUTL_VOL - [2:0] */
+
+/*
+ * R50 (0x32) - Output Mixer6
+ */
+#define WM8993_DACR_MIXOUTR_VOL_MASK 0x0E00 /* DACR_MIXOUTR_VOL - [11:9] */
+#define WM8993_DACR_MIXOUTR_VOL_SHIFT 9 /* DACR_MIXOUTR_VOL - [11:9] */
+#define WM8993_DACR_MIXOUTR_VOL_WIDTH 3 /* DACR_MIXOUTR_VOL - [11:9] */
+#define WM8993_IN2LN_MIXOUTR_VOL_MASK 0x01C0 /* IN2LN_MIXOUTR_VOL - [8:6] */
+#define WM8993_IN2LN_MIXOUTR_VOL_SHIFT 6 /* IN2LN_MIXOUTR_VOL - [8:6] */
+#define WM8993_IN2LN_MIXOUTR_VOL_WIDTH 3 /* IN2LN_MIXOUTR_VOL - [8:6] */
+#define WM8993_MIXINL_MIXOUTR_VOL_MASK 0x0038 /* MIXINL_MIXOUTR_VOL - [5:3] */
+#define WM8993_MIXINL_MIXOUTR_VOL_SHIFT 3 /* MIXINL_MIXOUTR_VOL - [5:3] */
+#define WM8993_MIXINL_MIXOUTR_VOL_WIDTH 3 /* MIXINL_MIXOUTR_VOL - [5:3] */
+#define WM8993_MIXINR_MIXOUTR_VOL_MASK 0x0007 /* MIXINR_MIXOUTR_VOL - [2:0] */
+#define WM8993_MIXINR_MIXOUTR_VOL_SHIFT 0 /* MIXINR_MIXOUTR_VOL - [2:0] */
+#define WM8993_MIXINR_MIXOUTR_VOL_WIDTH 3 /* MIXINR_MIXOUTR_VOL - [2:0] */
+
+/*
+ * R51 (0x33) - HPOUT2 Mixer
+ */
+#define WM8993_VRX_TO_HPOUT2 0x0020 /* VRX_TO_HPOUT2 */
+#define WM8993_VRX_TO_HPOUT2_MASK 0x0020 /* VRX_TO_HPOUT2 */
+#define WM8993_VRX_TO_HPOUT2_SHIFT 5 /* VRX_TO_HPOUT2 */
+#define WM8993_VRX_TO_HPOUT2_WIDTH 1 /* VRX_TO_HPOUT2 */
+#define WM8993_MIXOUTLVOL_TO_HPOUT2 0x0010 /* MIXOUTLVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTLVOL_TO_HPOUT2_MASK 0x0010 /* MIXOUTLVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTLVOL_TO_HPOUT2_SHIFT 4 /* MIXOUTLVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTLVOL_TO_HPOUT2_WIDTH 1 /* MIXOUTLVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTRVOL_TO_HPOUT2 0x0008 /* MIXOUTRVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTRVOL_TO_HPOUT2_MASK 0x0008 /* MIXOUTRVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTRVOL_TO_HPOUT2_SHIFT 3 /* MIXOUTRVOL_TO_HPOUT2 */
+#define WM8993_MIXOUTRVOL_TO_HPOUT2_WIDTH 1 /* MIXOUTRVOL_TO_HPOUT2 */
+
+/*
+ * R52 (0x34) - Line Mixer1
+ */
+#define WM8993_MIXOUTL_TO_LINEOUT1N 0x0040 /* MIXOUTL_TO_LINEOUT1N */
+#define WM8993_MIXOUTL_TO_LINEOUT1N_MASK 0x0040 /* MIXOUTL_TO_LINEOUT1N */
+#define WM8993_MIXOUTL_TO_LINEOUT1N_SHIFT 6 /* MIXOUTL_TO_LINEOUT1N */
+#define WM8993_MIXOUTL_TO_LINEOUT1N_WIDTH 1 /* MIXOUTL_TO_LINEOUT1N */
+#define WM8993_MIXOUTR_TO_LINEOUT1N 0x0020 /* MIXOUTR_TO_LINEOUT1N */
+#define WM8993_MIXOUTR_TO_LINEOUT1N_MASK 0x0020 /* MIXOUTR_TO_LINEOUT1N */
+#define WM8993_MIXOUTR_TO_LINEOUT1N_SHIFT 5 /* MIXOUTR_TO_LINEOUT1N */
+#define WM8993_MIXOUTR_TO_LINEOUT1N_WIDTH 1 /* MIXOUTR_TO_LINEOUT1N */
+#define WM8993_LINEOUT1_MODE 0x0010 /* LINEOUT1_MODE */
+#define WM8993_LINEOUT1_MODE_MASK 0x0010 /* LINEOUT1_MODE */
+#define WM8993_LINEOUT1_MODE_SHIFT 4 /* LINEOUT1_MODE */
+#define WM8993_LINEOUT1_MODE_WIDTH 1 /* LINEOUT1_MODE */
+#define WM8993_IN1R_TO_LINEOUT1P 0x0004 /* IN1R_TO_LINEOUT1P */
+#define WM8993_IN1R_TO_LINEOUT1P_MASK 0x0004 /* IN1R_TO_LINEOUT1P */
+#define WM8993_IN1R_TO_LINEOUT1P_SHIFT 2 /* IN1R_TO_LINEOUT1P */
+#define WM8993_IN1R_TO_LINEOUT1P_WIDTH 1 /* IN1R_TO_LINEOUT1P */
+#define WM8993_IN1L_TO_LINEOUT1P 0x0002 /* IN1L_TO_LINEOUT1P */
+#define WM8993_IN1L_TO_LINEOUT1P_MASK 0x0002 /* IN1L_TO_LINEOUT1P */
+#define WM8993_IN1L_TO_LINEOUT1P_SHIFT 1 /* IN1L_TO_LINEOUT1P */
+#define WM8993_IN1L_TO_LINEOUT1P_WIDTH 1 /* IN1L_TO_LINEOUT1P */
+#define WM8993_MIXOUTL_TO_LINEOUT1P 0x0001 /* MIXOUTL_TO_LINEOUT1P */
+#define WM8993_MIXOUTL_TO_LINEOUT1P_MASK 0x0001 /* MIXOUTL_TO_LINEOUT1P */
+#define WM8993_MIXOUTL_TO_LINEOUT1P_SHIFT 0 /* MIXOUTL_TO_LINEOUT1P */
+#define WM8993_MIXOUTL_TO_LINEOUT1P_WIDTH 1 /* MIXOUTL_TO_LINEOUT1P */
+
+/*
+ * R53 (0x35) - Line Mixer2
+ */
+#define WM8993_MIXOUTR_TO_LINEOUT2N 0x0040 /* MIXOUTR_TO_LINEOUT2N */
+#define WM8993_MIXOUTR_TO_LINEOUT2N_MASK 0x0040 /* MIXOUTR_TO_LINEOUT2N */
+#define WM8993_MIXOUTR_TO_LINEOUT2N_SHIFT 6 /* MIXOUTR_TO_LINEOUT2N */
+#define WM8993_MIXOUTR_TO_LINEOUT2N_WIDTH 1 /* MIXOUTR_TO_LINEOUT2N */
+#define WM8993_MIXOUTL_TO_LINEOUT2N 0x0020 /* MIXOUTL_TO_LINEOUT2N */
+#define WM8993_MIXOUTL_TO_LINEOUT2N_MASK 0x0020 /* MIXOUTL_TO_LINEOUT2N */
+#define WM8993_MIXOUTL_TO_LINEOUT2N_SHIFT 5 /* MIXOUTL_TO_LINEOUT2N */
+#define WM8993_MIXOUTL_TO_LINEOUT2N_WIDTH 1 /* MIXOUTL_TO_LINEOUT2N */
+#define WM8993_LINEOUT2_MODE 0x0010 /* LINEOUT2_MODE */
+#define WM8993_LINEOUT2_MODE_MASK 0x0010 /* LINEOUT2_MODE */
+#define WM8993_LINEOUT2_MODE_SHIFT 4 /* LINEOUT2_MODE */
+#define WM8993_LINEOUT2_MODE_WIDTH 1 /* LINEOUT2_MODE */
+#define WM8993_IN1L_TO_LINEOUT2P 0x0004 /* IN1L_TO_LINEOUT2P */
+#define WM8993_IN1L_TO_LINEOUT2P_MASK 0x0004 /* IN1L_TO_LINEOUT2P */
+#define WM8993_IN1L_TO_LINEOUT2P_SHIFT 2 /* IN1L_TO_LINEOUT2P */
+#define WM8993_IN1L_TO_LINEOUT2P_WIDTH 1 /* IN1L_TO_LINEOUT2P */
+#define WM8993_IN1R_TO_LINEOUT2P 0x0002 /* IN1R_TO_LINEOUT2P */
+#define WM8993_IN1R_TO_LINEOUT2P_MASK 0x0002 /* IN1R_TO_LINEOUT2P */
+#define WM8993_IN1R_TO_LINEOUT2P_SHIFT 1 /* IN1R_TO_LINEOUT2P */
+#define WM8993_IN1R_TO_LINEOUT2P_WIDTH 1 /* IN1R_TO_LINEOUT2P */
+#define WM8993_MIXOUTR_TO_LINEOUT2P 0x0001 /* MIXOUTR_TO_LINEOUT2P */
+#define WM8993_MIXOUTR_TO_LINEOUT2P_MASK 0x0001 /* MIXOUTR_TO_LINEOUT2P */
+#define WM8993_MIXOUTR_TO_LINEOUT2P_SHIFT 0 /* MIXOUTR_TO_LINEOUT2P */
+#define WM8993_MIXOUTR_TO_LINEOUT2P_WIDTH 1 /* MIXOUTR_TO_LINEOUT2P */
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM8993_SPKAB_REF_SEL 0x0100 /* SPKAB_REF_SEL */
+#define WM8993_SPKAB_REF_SEL_MASK 0x0100 /* SPKAB_REF_SEL */
+#define WM8993_SPKAB_REF_SEL_SHIFT 8 /* SPKAB_REF_SEL */
+#define WM8993_SPKAB_REF_SEL_WIDTH 1 /* SPKAB_REF_SEL */
+#define WM8993_MIXINL_TO_SPKMIXL 0x0080 /* MIXINL_TO_SPKMIXL */
+#define WM8993_MIXINL_TO_SPKMIXL_MASK 0x0080 /* MIXINL_TO_SPKMIXL */
+#define WM8993_MIXINL_TO_SPKMIXL_SHIFT 7 /* MIXINL_TO_SPKMIXL */
+#define WM8993_MIXINL_TO_SPKMIXL_WIDTH 1 /* MIXINL_TO_SPKMIXL */
+#define WM8993_MIXINR_TO_SPKMIXR 0x0040 /* MIXINR_TO_SPKMIXR */
+#define WM8993_MIXINR_TO_SPKMIXR_MASK 0x0040 /* MIXINR_TO_SPKMIXR */
+#define WM8993_MIXINR_TO_SPKMIXR_SHIFT 6 /* MIXINR_TO_SPKMIXR */
+#define WM8993_MIXINR_TO_SPKMIXR_WIDTH 1 /* MIXINR_TO_SPKMIXR */
+#define WM8993_IN1LP_TO_SPKMIXL 0x0020 /* IN1LP_TO_SPKMIXL */
+#define WM8993_IN1LP_TO_SPKMIXL_MASK 0x0020 /* IN1LP_TO_SPKMIXL */
+#define WM8993_IN1LP_TO_SPKMIXL_SHIFT 5 /* IN1LP_TO_SPKMIXL */
+#define WM8993_IN1LP_TO_SPKMIXL_WIDTH 1 /* IN1LP_TO_SPKMIXL */
+#define WM8993_IN1RP_TO_SPKMIXR 0x0010 /* IN1RP_TO_SPKMIXR */
+#define WM8993_IN1RP_TO_SPKMIXR_MASK 0x0010 /* IN1RP_TO_SPKMIXR */
+#define WM8993_IN1RP_TO_SPKMIXR_SHIFT 4 /* IN1RP_TO_SPKMIXR */
+#define WM8993_IN1RP_TO_SPKMIXR_WIDTH 1 /* IN1RP_TO_SPKMIXR */
+#define WM8993_MIXOUTL_TO_SPKMIXL 0x0008 /* MIXOUTL_TO_SPKMIXL */
+#define WM8993_MIXOUTL_TO_SPKMIXL_MASK 0x0008 /* MIXOUTL_TO_SPKMIXL */
+#define WM8993_MIXOUTL_TO_SPKMIXL_SHIFT 3 /* MIXOUTL_TO_SPKMIXL */
+#define WM8993_MIXOUTL_TO_SPKMIXL_WIDTH 1 /* MIXOUTL_TO_SPKMIXL */
+#define WM8993_MIXOUTR_TO_SPKMIXR 0x0004 /* MIXOUTR_TO_SPKMIXR */
+#define WM8993_MIXOUTR_TO_SPKMIXR_MASK 0x0004 /* MIXOUTR_TO_SPKMIXR */
+#define WM8993_MIXOUTR_TO_SPKMIXR_SHIFT 2 /* MIXOUTR_TO_SPKMIXR */
+#define WM8993_MIXOUTR_TO_SPKMIXR_WIDTH 1 /* MIXOUTR_TO_SPKMIXR */
+#define WM8993_DACL_TO_SPKMIXL 0x0002 /* DACL_TO_SPKMIXL */
+#define WM8993_DACL_TO_SPKMIXL_MASK 0x0002 /* DACL_TO_SPKMIXL */
+#define WM8993_DACL_TO_SPKMIXL_SHIFT 1 /* DACL_TO_SPKMIXL */
+#define WM8993_DACL_TO_SPKMIXL_WIDTH 1 /* DACL_TO_SPKMIXL */
+#define WM8993_DACR_TO_SPKMIXR 0x0001 /* DACR_TO_SPKMIXR */
+#define WM8993_DACR_TO_SPKMIXR_MASK 0x0001 /* DACR_TO_SPKMIXR */
+#define WM8993_DACR_TO_SPKMIXR_SHIFT 0 /* DACR_TO_SPKMIXR */
+#define WM8993_DACR_TO_SPKMIXR_WIDTH 1 /* DACR_TO_SPKMIXR */
+
+/*
+ * R55 (0x37) - Additional Control
+ */
+#define WM8993_LINEOUT1_FB 0x0080 /* LINEOUT1_FB */
+#define WM8993_LINEOUT1_FB_MASK 0x0080 /* LINEOUT1_FB */
+#define WM8993_LINEOUT1_FB_SHIFT 7 /* LINEOUT1_FB */
+#define WM8993_LINEOUT1_FB_WIDTH 1 /* LINEOUT1_FB */
+#define WM8993_LINEOUT2_FB 0x0040 /* LINEOUT2_FB */
+#define WM8993_LINEOUT2_FB_MASK 0x0040 /* LINEOUT2_FB */
+#define WM8993_LINEOUT2_FB_SHIFT 6 /* LINEOUT2_FB */
+#define WM8993_LINEOUT2_FB_WIDTH 1 /* LINEOUT2_FB */
+#define WM8993_VROI 0x0001 /* VROI */
+#define WM8993_VROI_MASK 0x0001 /* VROI */
+#define WM8993_VROI_SHIFT 0 /* VROI */
+#define WM8993_VROI_WIDTH 1 /* VROI */
+
+/*
+ * R56 (0x38) - AntiPOP1
+ */
+#define WM8993_LINEOUT_VMID_BUF_ENA 0x0080 /* LINEOUT_VMID_BUF_ENA */
+#define WM8993_LINEOUT_VMID_BUF_ENA_MASK 0x0080 /* LINEOUT_VMID_BUF_ENA */
+#define WM8993_LINEOUT_VMID_BUF_ENA_SHIFT 7 /* LINEOUT_VMID_BUF_ENA */
+#define WM8993_LINEOUT_VMID_BUF_ENA_WIDTH 1 /* LINEOUT_VMID_BUF_ENA */
+#define WM8993_HPOUT2_IN_ENA 0x0040 /* HPOUT2_IN_ENA */
+#define WM8993_HPOUT2_IN_ENA_MASK 0x0040 /* HPOUT2_IN_ENA */
+#define WM8993_HPOUT2_IN_ENA_SHIFT 6 /* HPOUT2_IN_ENA */
+#define WM8993_HPOUT2_IN_ENA_WIDTH 1 /* HPOUT2_IN_ENA */
+#define WM8993_LINEOUT1_DISCH 0x0020 /* LINEOUT1_DISCH */
+#define WM8993_LINEOUT1_DISCH_MASK 0x0020 /* LINEOUT1_DISCH */
+#define WM8993_LINEOUT1_DISCH_SHIFT 5 /* LINEOUT1_DISCH */
+#define WM8993_LINEOUT1_DISCH_WIDTH 1 /* LINEOUT1_DISCH */
+#define WM8993_LINEOUT2_DISCH 0x0010 /* LINEOUT2_DISCH */
+#define WM8993_LINEOUT2_DISCH_MASK 0x0010 /* LINEOUT2_DISCH */
+#define WM8993_LINEOUT2_DISCH_SHIFT 4 /* LINEOUT2_DISCH */
+#define WM8993_LINEOUT2_DISCH_WIDTH 1 /* LINEOUT2_DISCH */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM8993_VMID_RAMP_MASK 0x0060 /* VMID_RAMP - [6:5] */
+#define WM8993_VMID_RAMP_SHIFT 5 /* VMID_RAMP - [6:5] */
+#define WM8993_VMID_RAMP_WIDTH 2 /* VMID_RAMP - [6:5] */
+#define WM8993_VMID_BUF_ENA 0x0008 /* VMID_BUF_ENA */
+#define WM8993_VMID_BUF_ENA_MASK 0x0008 /* VMID_BUF_ENA */
+#define WM8993_VMID_BUF_ENA_SHIFT 3 /* VMID_BUF_ENA */
+#define WM8993_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */
+#define WM8993_STARTUP_BIAS_ENA 0x0004 /* STARTUP_BIAS_ENA */
+#define WM8993_STARTUP_BIAS_ENA_MASK 0x0004 /* STARTUP_BIAS_ENA */
+#define WM8993_STARTUP_BIAS_ENA_SHIFT 2 /* STARTUP_BIAS_ENA */
+#define WM8993_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */
+#define WM8993_BIAS_SRC 0x0002 /* BIAS_SRC */
+#define WM8993_BIAS_SRC_MASK 0x0002 /* BIAS_SRC */
+#define WM8993_BIAS_SRC_SHIFT 1 /* BIAS_SRC */
+#define WM8993_BIAS_SRC_WIDTH 1 /* BIAS_SRC */
+#define WM8993_VMID_DISCH 0x0001 /* VMID_DISCH */
+#define WM8993_VMID_DISCH_MASK 0x0001 /* VMID_DISCH */
+#define WM8993_VMID_DISCH_SHIFT 0 /* VMID_DISCH */
+#define WM8993_VMID_DISCH_WIDTH 1 /* VMID_DISCH */
+
+/*
+ * R58 (0x3A) - MICBIAS
+ */
+#define WM8993_JD_SCTHR_MASK 0x00C0 /* JD_SCTHR - [7:6] */
+#define WM8993_JD_SCTHR_SHIFT 6 /* JD_SCTHR - [7:6] */
+#define WM8993_JD_SCTHR_WIDTH 2 /* JD_SCTHR - [7:6] */
+#define WM8993_JD_THR_MASK 0x0030 /* JD_THR - [5:4] */
+#define WM8993_JD_THR_SHIFT 4 /* JD_THR - [5:4] */
+#define WM8993_JD_THR_WIDTH 2 /* JD_THR - [5:4] */
+#define WM8993_JD_ENA 0x0004 /* JD_ENA */
+#define WM8993_JD_ENA_MASK 0x0004 /* JD_ENA */
+#define WM8993_JD_ENA_SHIFT 2 /* JD_ENA */
+#define WM8993_JD_ENA_WIDTH 1 /* JD_ENA */
+#define WM8993_MICB2_LVL 0x0002 /* MICB2_LVL */
+#define WM8993_MICB2_LVL_MASK 0x0002 /* MICB2_LVL */
+#define WM8993_MICB2_LVL_SHIFT 1 /* MICB2_LVL */
+#define WM8993_MICB2_LVL_WIDTH 1 /* MICB2_LVL */
+#define WM8993_MICB1_LVL 0x0001 /* MICB1_LVL */
+#define WM8993_MICB1_LVL_MASK 0x0001 /* MICB1_LVL */
+#define WM8993_MICB1_LVL_SHIFT 0 /* MICB1_LVL */
+#define WM8993_MICB1_LVL_WIDTH 1 /* MICB1_LVL */
+
+/*
+ * R60 (0x3C) - FLL Control 1
+ */
+#define WM8993_FLL_FRAC 0x0004 /* FLL_FRAC */
+#define WM8993_FLL_FRAC_MASK 0x0004 /* FLL_FRAC */
+#define WM8993_FLL_FRAC_SHIFT 2 /* FLL_FRAC */
+#define WM8993_FLL_FRAC_WIDTH 1 /* FLL_FRAC */
+#define WM8993_FLL_OSC_ENA 0x0002 /* FLL_OSC_ENA */
+#define WM8993_FLL_OSC_ENA_MASK 0x0002 /* FLL_OSC_ENA */
+#define WM8993_FLL_OSC_ENA_SHIFT 1 /* FLL_OSC_ENA */
+#define WM8993_FLL_OSC_ENA_WIDTH 1 /* FLL_OSC_ENA */
+#define WM8993_FLL_ENA 0x0001 /* FLL_ENA */
+#define WM8993_FLL_ENA_MASK 0x0001 /* FLL_ENA */
+#define WM8993_FLL_ENA_SHIFT 0 /* FLL_ENA */
+#define WM8993_FLL_ENA_WIDTH 1 /* FLL_ENA */
+
+/*
+ * R61 (0x3D) - FLL Control 2
+ */
+#define WM8993_FLL_OUTDIV_MASK 0x0700 /* FLL_OUTDIV - [10:8] */
+#define WM8993_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [10:8] */
+#define WM8993_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [10:8] */
+#define WM8993_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */
+#define WM8993_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */
+#define WM8993_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */
+#define WM8993_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */
+#define WM8993_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */
+#define WM8993_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */
+
+/*
+ * R62 (0x3E) - FLL Control 3
+ */
+#define WM8993_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */
+#define WM8993_FLL_K_SHIFT 0 /* FLL_K - [15:0] */
+#define WM8993_FLL_K_WIDTH 16 /* FLL_K - [15:0] */
+
+/*
+ * R63 (0x3F) - FLL Control 4
+ */
+#define WM8993_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */
+#define WM8993_FLL_N_SHIFT 5 /* FLL_N - [14:5] */
+#define WM8993_FLL_N_WIDTH 10 /* FLL_N - [14:5] */
+#define WM8993_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */
+#define WM8993_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */
+#define WM8993_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */
+
+/*
+ * R64 (0x40) - FLL Control 5
+ */
+#define WM8993_FLL_FRC_NCO_VAL_MASK 0x1F80 /* FLL_FRC_NCO_VAL - [12:7] */
+#define WM8993_FLL_FRC_NCO_VAL_SHIFT 7 /* FLL_FRC_NCO_VAL - [12:7] */
+#define WM8993_FLL_FRC_NCO_VAL_WIDTH 6 /* FLL_FRC_NCO_VAL - [12:7] */
+#define WM8993_FLL_FRC_NCO 0x0040 /* FLL_FRC_NCO */
+#define WM8993_FLL_FRC_NCO_MASK 0x0040 /* FLL_FRC_NCO */
+#define WM8993_FLL_FRC_NCO_SHIFT 6 /* FLL_FRC_NCO */
+#define WM8993_FLL_FRC_NCO_WIDTH 1 /* FLL_FRC_NCO */
+#define WM8993_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM8993_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM8993_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM8993_FLL_CLK_SRC_MASK 0x0003 /* FLL_CLK_SRC - [1:0] */
+#define WM8993_FLL_CLK_SRC_SHIFT 0 /* FLL_CLK_SRC - [1:0] */
+#define WM8993_FLL_CLK_SRC_WIDTH 2 /* FLL_CLK_SRC - [1:0] */
+
+/*
+ * R65 (0x41) - Clocking 3
+ */
+#define WM8993_CLK_DCS_DIV_MASK 0x3C00 /* CLK_DCS_DIV - [13:10] */
+#define WM8993_CLK_DCS_DIV_SHIFT 10 /* CLK_DCS_DIV - [13:10] */
+#define WM8993_CLK_DCS_DIV_WIDTH 4 /* CLK_DCS_DIV - [13:10] */
+#define WM8993_SAMPLE_RATE_MASK 0x0380 /* SAMPLE_RATE - [9:7] */
+#define WM8993_SAMPLE_RATE_SHIFT 7 /* SAMPLE_RATE - [9:7] */
+#define WM8993_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [9:7] */
+#define WM8993_CLK_SYS_RATE_MASK 0x001E /* CLK_SYS_RATE - [4:1] */
+#define WM8993_CLK_SYS_RATE_SHIFT 1 /* CLK_SYS_RATE - [4:1] */
+#define WM8993_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [4:1] */
+#define WM8993_CLK_DSP_ENA 0x0001 /* CLK_DSP_ENA */
+#define WM8993_CLK_DSP_ENA_MASK 0x0001 /* CLK_DSP_ENA */
+#define WM8993_CLK_DSP_ENA_SHIFT 0 /* CLK_DSP_ENA */
+#define WM8993_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */
+
+/*
+ * R66 (0x42) - Clocking 4
+ */
+#define WM8993_DAC_DIV4 0x0200 /* DAC_DIV4 */
+#define WM8993_DAC_DIV4_MASK 0x0200 /* DAC_DIV4 */
+#define WM8993_DAC_DIV4_SHIFT 9 /* DAC_DIV4 */
+#define WM8993_DAC_DIV4_WIDTH 1 /* DAC_DIV4 */
+#define WM8993_CLK_256K_DIV_MASK 0x007E /* CLK_256K_DIV - [6:1] */
+#define WM8993_CLK_256K_DIV_SHIFT 1 /* CLK_256K_DIV - [6:1] */
+#define WM8993_CLK_256K_DIV_WIDTH 6 /* CLK_256K_DIV - [6:1] */
+#define WM8993_SR_MODE 0x0001 /* SR_MODE */
+#define WM8993_SR_MODE_MASK 0x0001 /* SR_MODE */
+#define WM8993_SR_MODE_SHIFT 0 /* SR_MODE */
+#define WM8993_SR_MODE_WIDTH 1 /* SR_MODE */
+
+/*
+ * R67 (0x43) - MW Slave Control
+ */
+#define WM8993_MASK_WRITE_ENA 0x0001 /* MASK_WRITE_ENA */
+#define WM8993_MASK_WRITE_ENA_MASK 0x0001 /* MASK_WRITE_ENA */
+#define WM8993_MASK_WRITE_ENA_SHIFT 0 /* MASK_WRITE_ENA */
+#define WM8993_MASK_WRITE_ENA_WIDTH 1 /* MASK_WRITE_ENA */
+
+/*
+ * R69 (0x45) - Bus Control 1
+ */
+#define WM8993_CLK_SYS_ENA 0x0002 /* CLK_SYS_ENA */
+#define WM8993_CLK_SYS_ENA_MASK 0x0002 /* CLK_SYS_ENA */
+#define WM8993_CLK_SYS_ENA_SHIFT 1 /* CLK_SYS_ENA */
+#define WM8993_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */
+
+/*
+ * R70 (0x46) - Write Sequencer 0
+ */
+#define WM8993_WSEQ_ENA 0x0100 /* WSEQ_ENA */
+#define WM8993_WSEQ_ENA_MASK 0x0100 /* WSEQ_ENA */
+#define WM8993_WSEQ_ENA_SHIFT 8 /* WSEQ_ENA */
+#define WM8993_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */
+#define WM8993_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8993_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8993_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */
+
+/*
+ * R71 (0x47) - Write Sequencer 1
+ */
+#define WM8993_WSEQ_DATA_WIDTH_MASK 0x7000 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8993_WSEQ_DATA_WIDTH_SHIFT 12 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8993_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8993_WSEQ_DATA_START_MASK 0x0F00 /* WSEQ_DATA_START - [11:8] */
+#define WM8993_WSEQ_DATA_START_SHIFT 8 /* WSEQ_DATA_START - [11:8] */
+#define WM8993_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [11:8] */
+#define WM8993_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */
+#define WM8993_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */
+#define WM8993_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */
+
+/*
+ * R72 (0x48) - Write Sequencer 2
+ */
+#define WM8993_WSEQ_EOS 0x4000 /* WSEQ_EOS */
+#define WM8993_WSEQ_EOS_MASK 0x4000 /* WSEQ_EOS */
+#define WM8993_WSEQ_EOS_SHIFT 14 /* WSEQ_EOS */
+#define WM8993_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */
+#define WM8993_WSEQ_DELAY_MASK 0x0F00 /* WSEQ_DELAY - [11:8] */
+#define WM8993_WSEQ_DELAY_SHIFT 8 /* WSEQ_DELAY - [11:8] */
+#define WM8993_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [11:8] */
+#define WM8993_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */
+#define WM8993_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */
+#define WM8993_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */
+
+/*
+ * R73 (0x49) - Write Sequencer 3
+ */
+#define WM8993_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */
+#define WM8993_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */
+#define WM8993_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */
+#define WM8993_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */
+#define WM8993_WSEQ_START 0x0100 /* WSEQ_START */
+#define WM8993_WSEQ_START_MASK 0x0100 /* WSEQ_START */
+#define WM8993_WSEQ_START_SHIFT 8 /* WSEQ_START */
+#define WM8993_WSEQ_START_WIDTH 1 /* WSEQ_START */
+#define WM8993_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */
+#define WM8993_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */
+#define WM8993_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */
+
+/*
+ * R74 (0x4A) - Write Sequencer 4
+ */
+#define WM8993_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */
+#define WM8993_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */
+#define WM8993_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */
+#define WM8993_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */
+
+/*
+ * R75 (0x4B) - Write Sequencer 5
+ */
+#define WM8993_WSEQ_CURRENT_INDEX_MASK 0x003F /* WSEQ_CURRENT_INDEX - [5:0] */
+#define WM8993_WSEQ_CURRENT_INDEX_SHIFT 0 /* WSEQ_CURRENT_INDEX - [5:0] */
+#define WM8993_WSEQ_CURRENT_INDEX_WIDTH 6 /* WSEQ_CURRENT_INDEX - [5:0] */
+
+/*
+ * R76 (0x4C) - Charge Pump 1
+ */
+#define WM8993_CP_ENA 0x8000 /* CP_ENA */
+#define WM8993_CP_ENA_MASK 0x8000 /* CP_ENA */
+#define WM8993_CP_ENA_SHIFT 15 /* CP_ENA */
+#define WM8993_CP_ENA_WIDTH 1 /* CP_ENA */
+
+/*
+ * R81 (0x51) - Class W 0
+ */
+#define WM8993_CP_DYN_FREQ 0x0002 /* CP_DYN_FREQ */
+#define WM8993_CP_DYN_FREQ_MASK 0x0002 /* CP_DYN_FREQ */
+#define WM8993_CP_DYN_FREQ_SHIFT 1 /* CP_DYN_FREQ */
+#define WM8993_CP_DYN_FREQ_WIDTH 1 /* CP_DYN_FREQ */
+#define WM8993_CP_DYN_V 0x0001 /* CP_DYN_V */
+#define WM8993_CP_DYN_V_MASK 0x0001 /* CP_DYN_V */
+#define WM8993_CP_DYN_V_SHIFT 0 /* CP_DYN_V */
+#define WM8993_CP_DYN_V_WIDTH 1 /* CP_DYN_V */
+
+/*
+ * R84 (0x54) - DC Servo 0
+ */
+#define WM8993_DCS_TRIG_SINGLE_1 0x2000 /* DCS_TRIG_SINGLE_1 */
+#define WM8993_DCS_TRIG_SINGLE_1_MASK 0x2000 /* DCS_TRIG_SINGLE_1 */
+#define WM8993_DCS_TRIG_SINGLE_1_SHIFT 13 /* DCS_TRIG_SINGLE_1 */
+#define WM8993_DCS_TRIG_SINGLE_1_WIDTH 1 /* DCS_TRIG_SINGLE_1 */
+#define WM8993_DCS_TRIG_SINGLE_0 0x1000 /* DCS_TRIG_SINGLE_0 */
+#define WM8993_DCS_TRIG_SINGLE_0_MASK 0x1000 /* DCS_TRIG_SINGLE_0 */
+#define WM8993_DCS_TRIG_SINGLE_0_SHIFT 12 /* DCS_TRIG_SINGLE_0 */
+#define WM8993_DCS_TRIG_SINGLE_0_WIDTH 1 /* DCS_TRIG_SINGLE_0 */
+#define WM8993_DCS_TRIG_SERIES_1 0x0200 /* DCS_TRIG_SERIES_1 */
+#define WM8993_DCS_TRIG_SERIES_1_MASK 0x0200 /* DCS_TRIG_SERIES_1 */
+#define WM8993_DCS_TRIG_SERIES_1_SHIFT 9 /* DCS_TRIG_SERIES_1 */
+#define WM8993_DCS_TRIG_SERIES_1_WIDTH 1 /* DCS_TRIG_SERIES_1 */
+#define WM8993_DCS_TRIG_SERIES_0 0x0100 /* DCS_TRIG_SERIES_0 */
+#define WM8993_DCS_TRIG_SERIES_0_MASK 0x0100 /* DCS_TRIG_SERIES_0 */
+#define WM8993_DCS_TRIG_SERIES_0_SHIFT 8 /* DCS_TRIG_SERIES_0 */
+#define WM8993_DCS_TRIG_SERIES_0_WIDTH 1 /* DCS_TRIG_SERIES_0 */
+#define WM8993_DCS_TRIG_STARTUP_1 0x0020 /* DCS_TRIG_STARTUP_1 */
+#define WM8993_DCS_TRIG_STARTUP_1_MASK 0x0020 /* DCS_TRIG_STARTUP_1 */
+#define WM8993_DCS_TRIG_STARTUP_1_SHIFT 5 /* DCS_TRIG_STARTUP_1 */
+#define WM8993_DCS_TRIG_STARTUP_1_WIDTH 1 /* DCS_TRIG_STARTUP_1 */
+#define WM8993_DCS_TRIG_STARTUP_0 0x0010 /* DCS_TRIG_STARTUP_0 */
+#define WM8993_DCS_TRIG_STARTUP_0_MASK 0x0010 /* DCS_TRIG_STARTUP_0 */
+#define WM8993_DCS_TRIG_STARTUP_0_SHIFT 4 /* DCS_TRIG_STARTUP_0 */
+#define WM8993_DCS_TRIG_STARTUP_0_WIDTH 1 /* DCS_TRIG_STARTUP_0 */
+#define WM8993_DCS_TRIG_DAC_WR_1 0x0008 /* DCS_TRIG_DAC_WR_1 */
+#define WM8993_DCS_TRIG_DAC_WR_1_MASK 0x0008 /* DCS_TRIG_DAC_WR_1 */
+#define WM8993_DCS_TRIG_DAC_WR_1_SHIFT 3 /* DCS_TRIG_DAC_WR_1 */
+#define WM8993_DCS_TRIG_DAC_WR_1_WIDTH 1 /* DCS_TRIG_DAC_WR_1 */
+#define WM8993_DCS_TRIG_DAC_WR_0 0x0004 /* DCS_TRIG_DAC_WR_0 */
+#define WM8993_DCS_TRIG_DAC_WR_0_MASK 0x0004 /* DCS_TRIG_DAC_WR_0 */
+#define WM8993_DCS_TRIG_DAC_WR_0_SHIFT 2 /* DCS_TRIG_DAC_WR_0 */
+#define WM8993_DCS_TRIG_DAC_WR_0_WIDTH 1 /* DCS_TRIG_DAC_WR_0 */
+#define WM8993_DCS_ENA_CHAN_1 0x0002 /* DCS_ENA_CHAN_1 */
+#define WM8993_DCS_ENA_CHAN_1_MASK 0x0002 /* DCS_ENA_CHAN_1 */
+#define WM8993_DCS_ENA_CHAN_1_SHIFT 1 /* DCS_ENA_CHAN_1 */
+#define WM8993_DCS_ENA_CHAN_1_WIDTH 1 /* DCS_ENA_CHAN_1 */
+#define WM8993_DCS_ENA_CHAN_0 0x0001 /* DCS_ENA_CHAN_0 */
+#define WM8993_DCS_ENA_CHAN_0_MASK 0x0001 /* DCS_ENA_CHAN_0 */
+#define WM8993_DCS_ENA_CHAN_0_SHIFT 0 /* DCS_ENA_CHAN_0 */
+#define WM8993_DCS_ENA_CHAN_0_WIDTH 1 /* DCS_ENA_CHAN_0 */
+
+/*
+ * R85 (0x55) - DC Servo 1
+ */
+#define WM8993_DCS_SERIES_NO_01_MASK 0x0FE0 /* DCS_SERIES_NO_01 - [11:5] */
+#define WM8993_DCS_SERIES_NO_01_SHIFT 5 /* DCS_SERIES_NO_01 - [11:5] */
+#define WM8993_DCS_SERIES_NO_01_WIDTH 7 /* DCS_SERIES_NO_01 - [11:5] */
+#define WM8993_DCS_TIMER_PERIOD_01_MASK 0x000F /* DCS_TIMER_PERIOD_01 - [3:0] */
+#define WM8993_DCS_TIMER_PERIOD_01_SHIFT 0 /* DCS_TIMER_PERIOD_01 - [3:0] */
+#define WM8993_DCS_TIMER_PERIOD_01_WIDTH 4 /* DCS_TIMER_PERIOD_01 - [3:0] */
+
+/*
+ * R87 (0x57) - DC Servo 3
+ */
+#define WM8993_DCS_DAC_WR_VAL_1_MASK 0xFF00 /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM8993_DCS_DAC_WR_VAL_1_SHIFT 8 /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM8993_DCS_DAC_WR_VAL_1_WIDTH 8 /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM8993_DCS_DAC_WR_VAL_0_MASK 0x00FF /* DCS_DAC_WR_VAL_0 - [7:0] */
+#define WM8993_DCS_DAC_WR_VAL_0_SHIFT 0 /* DCS_DAC_WR_VAL_0 - [7:0] */
+#define WM8993_DCS_DAC_WR_VAL_0_WIDTH 8 /* DCS_DAC_WR_VAL_0 - [7:0] */
+
+/*
+ * R88 (0x58) - DC Servo Readback 0
+ */
+#define WM8993_DCS_DATAPATH_BUSY 0x4000 /* DCS_DATAPATH_BUSY */
+#define WM8993_DCS_DATAPATH_BUSY_MASK 0x4000 /* DCS_DATAPATH_BUSY */
+#define WM8993_DCS_DATAPATH_BUSY_SHIFT 14 /* DCS_DATAPATH_BUSY */
+#define WM8993_DCS_DATAPATH_BUSY_WIDTH 1 /* DCS_DATAPATH_BUSY */
+#define WM8993_DCS_CHANNEL_MASK 0x3000 /* DCS_CHANNEL - [13:12] */
+#define WM8993_DCS_CHANNEL_SHIFT 12 /* DCS_CHANNEL - [13:12] */
+#define WM8993_DCS_CHANNEL_WIDTH 2 /* DCS_CHANNEL - [13:12] */
+#define WM8993_DCS_CAL_COMPLETE_MASK 0x0300 /* DCS_CAL_COMPLETE - [9:8] */
+#define WM8993_DCS_CAL_COMPLETE_SHIFT 8 /* DCS_CAL_COMPLETE - [9:8] */
+#define WM8993_DCS_CAL_COMPLETE_WIDTH 2 /* DCS_CAL_COMPLETE - [9:8] */
+#define WM8993_DCS_DAC_WR_COMPLETE_MASK 0x0030 /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM8993_DCS_DAC_WR_COMPLETE_SHIFT 4 /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM8993_DCS_DAC_WR_COMPLETE_WIDTH 2 /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM8993_DCS_STARTUP_COMPLETE_MASK 0x0003 /* DCS_STARTUP_COMPLETE - [1:0] */
+#define WM8993_DCS_STARTUP_COMPLETE_SHIFT 0 /* DCS_STARTUP_COMPLETE - [1:0] */
+#define WM8993_DCS_STARTUP_COMPLETE_WIDTH 2 /* DCS_STARTUP_COMPLETE - [1:0] */
+
+/*
+ * R89 (0x59) - DC Servo Readback 1
+ */
+#define WM8993_DCS_INTEG_CHAN_1_MASK 0x00FF /* DCS_INTEG_CHAN_1 - [7:0] */
+#define WM8993_DCS_INTEG_CHAN_1_SHIFT 0 /* DCS_INTEG_CHAN_1 - [7:0] */
+#define WM8993_DCS_INTEG_CHAN_1_WIDTH 8 /* DCS_INTEG_CHAN_1 - [7:0] */
+
+/*
+ * R90 (0x5A) - DC Servo Readback 2
+ */
+#define WM8993_DCS_INTEG_CHAN_0_MASK 0x00FF /* DCS_INTEG_CHAN_0 - [7:0] */
+#define WM8993_DCS_INTEG_CHAN_0_SHIFT 0 /* DCS_INTEG_CHAN_0 - [7:0] */
+#define WM8993_DCS_INTEG_CHAN_0_WIDTH 8 /* DCS_INTEG_CHAN_0 - [7:0] */
+
+/*
+ * R96 (0x60) - Analogue HP 0
+ */
+#define WM8993_HPOUT1_AUTO_PU 0x0100 /* HPOUT1_AUTO_PU */
+#define WM8993_HPOUT1_AUTO_PU_MASK 0x0100 /* HPOUT1_AUTO_PU */
+#define WM8993_HPOUT1_AUTO_PU_SHIFT 8 /* HPOUT1_AUTO_PU */
+#define WM8993_HPOUT1_AUTO_PU_WIDTH 1 /* HPOUT1_AUTO_PU */
+#define WM8993_HPOUT1L_RMV_SHORT 0x0080 /* HPOUT1L_RMV_SHORT */
+#define WM8993_HPOUT1L_RMV_SHORT_MASK 0x0080 /* HPOUT1L_RMV_SHORT */
+#define WM8993_HPOUT1L_RMV_SHORT_SHIFT 7 /* HPOUT1L_RMV_SHORT */
+#define WM8993_HPOUT1L_RMV_SHORT_WIDTH 1 /* HPOUT1L_RMV_SHORT */
+#define WM8993_HPOUT1L_OUTP 0x0040 /* HPOUT1L_OUTP */
+#define WM8993_HPOUT1L_OUTP_MASK 0x0040 /* HPOUT1L_OUTP */
+#define WM8993_HPOUT1L_OUTP_SHIFT 6 /* HPOUT1L_OUTP */
+#define WM8993_HPOUT1L_OUTP_WIDTH 1 /* HPOUT1L_OUTP */
+#define WM8993_HPOUT1L_DLY 0x0020 /* HPOUT1L_DLY */
+#define WM8993_HPOUT1L_DLY_MASK 0x0020 /* HPOUT1L_DLY */
+#define WM8993_HPOUT1L_DLY_SHIFT 5 /* HPOUT1L_DLY */
+#define WM8993_HPOUT1L_DLY_WIDTH 1 /* HPOUT1L_DLY */
+#define WM8993_HPOUT1R_RMV_SHORT 0x0008 /* HPOUT1R_RMV_SHORT */
+#define WM8993_HPOUT1R_RMV_SHORT_MASK 0x0008 /* HPOUT1R_RMV_SHORT */
+#define WM8993_HPOUT1R_RMV_SHORT_SHIFT 3 /* HPOUT1R_RMV_SHORT */
+#define WM8993_HPOUT1R_RMV_SHORT_WIDTH 1 /* HPOUT1R_RMV_SHORT */
+#define WM8993_HPOUT1R_OUTP 0x0004 /* HPOUT1R_OUTP */
+#define WM8993_HPOUT1R_OUTP_MASK 0x0004 /* HPOUT1R_OUTP */
+#define WM8993_HPOUT1R_OUTP_SHIFT 2 /* HPOUT1R_OUTP */
+#define WM8993_HPOUT1R_OUTP_WIDTH 1 /* HPOUT1R_OUTP */
+#define WM8993_HPOUT1R_DLY 0x0002 /* HPOUT1R_DLY */
+#define WM8993_HPOUT1R_DLY_MASK 0x0002 /* HPOUT1R_DLY */
+#define WM8993_HPOUT1R_DLY_SHIFT 1 /* HPOUT1R_DLY */
+#define WM8993_HPOUT1R_DLY_WIDTH 1 /* HPOUT1R_DLY */
+
+/*
+ * R98 (0x62) - EQ1
+ */
+#define WM8993_EQ_ENA 0x0001 /* EQ_ENA */
+#define WM8993_EQ_ENA_MASK 0x0001 /* EQ_ENA */
+#define WM8993_EQ_ENA_SHIFT 0 /* EQ_ENA */
+#define WM8993_EQ_ENA_WIDTH 1 /* EQ_ENA */
+
+/*
+ * R99 (0x63) - EQ2
+ */
+#define WM8993_EQ_B1_GAIN_MASK 0x001F /* EQ_B1_GAIN - [4:0] */
+#define WM8993_EQ_B1_GAIN_SHIFT 0 /* EQ_B1_GAIN - [4:0] */
+#define WM8993_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [4:0] */
+
+/*
+ * R100 (0x64) - EQ3
+ */
+#define WM8993_EQ_B2_GAIN_MASK 0x001F /* EQ_B2_GAIN - [4:0] */
+#define WM8993_EQ_B2_GAIN_SHIFT 0 /* EQ_B2_GAIN - [4:0] */
+#define WM8993_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [4:0] */
+
+/*
+ * R101 (0x65) - EQ4
+ */
+#define WM8993_EQ_B3_GAIN_MASK 0x001F /* EQ_B3_GAIN - [4:0] */
+#define WM8993_EQ_B3_GAIN_SHIFT 0 /* EQ_B3_GAIN - [4:0] */
+#define WM8993_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [4:0] */
+
+/*
+ * R102 (0x66) - EQ5
+ */
+#define WM8993_EQ_B4_GAIN_MASK 0x001F /* EQ_B4_GAIN - [4:0] */
+#define WM8993_EQ_B4_GAIN_SHIFT 0 /* EQ_B4_GAIN - [4:0] */
+#define WM8993_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [4:0] */
+
+/*
+ * R103 (0x67) - EQ6
+ */
+#define WM8993_EQ_B5_GAIN_MASK 0x001F /* EQ_B5_GAIN - [4:0] */
+#define WM8993_EQ_B5_GAIN_SHIFT 0 /* EQ_B5_GAIN - [4:0] */
+#define WM8993_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [4:0] */
+
+/*
+ * R104 (0x68) - EQ7
+ */
+#define WM8993_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */
+#define WM8993_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */
+#define WM8993_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */
+
+/*
+ * R105 (0x69) - EQ8
+ */
+#define WM8993_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */
+#define WM8993_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */
+#define WM8993_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */
+
+/*
+ * R106 (0x6A) - EQ9
+ */
+#define WM8993_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */
+#define WM8993_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */
+#define WM8993_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */
+
+/*
+ * R107 (0x6B) - EQ10
+ */
+#define WM8993_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */
+#define WM8993_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */
+#define WM8993_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */
+
+/*
+ * R108 (0x6C) - EQ11
+ */
+#define WM8993_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */
+#define WM8993_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */
+#define WM8993_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */
+
+/*
+ * R109 (0x6D) - EQ12
+ */
+#define WM8993_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */
+#define WM8993_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */
+#define WM8993_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */
+
+/*
+ * R110 (0x6E) - EQ13
+ */
+#define WM8993_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */
+#define WM8993_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */
+#define WM8993_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */
+
+/*
+ * R111 (0x6F) - EQ14
+ */
+#define WM8993_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */
+#define WM8993_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */
+#define WM8993_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */
+
+/*
+ * R112 (0x70) - EQ15
+ */
+#define WM8993_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */
+#define WM8993_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */
+#define WM8993_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */
+
+/*
+ * R113 (0x71) - EQ16
+ */
+#define WM8993_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */
+#define WM8993_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */
+#define WM8993_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */
+
+/*
+ * R114 (0x72) - EQ17
+ */
+#define WM8993_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */
+#define WM8993_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */
+#define WM8993_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */
+
+/*
+ * R115 (0x73) - EQ18
+ */
+#define WM8993_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */
+#define WM8993_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */
+#define WM8993_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */
+
+/*
+ * R116 (0x74) - EQ19
+ */
+#define WM8993_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */
+#define WM8993_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */
+#define WM8993_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */
+
+/*
+ * R117 (0x75) - EQ20
+ */
+#define WM8993_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */
+#define WM8993_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */
+#define WM8993_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */
+
+/*
+ * R118 (0x76) - EQ21
+ */
+#define WM8993_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */
+#define WM8993_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */
+#define WM8993_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */
+
+/*
+ * R119 (0x77) - EQ22
+ */
+#define WM8993_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */
+#define WM8993_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */
+#define WM8993_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */
+
+/*
+ * R120 (0x78) - EQ23
+ */
+#define WM8993_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */
+#define WM8993_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */
+#define WM8993_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */
+
+/*
+ * R121 (0x79) - EQ24
+ */
+#define WM8993_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */
+#define WM8993_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */
+#define WM8993_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */
+
+/*
+ * R122 (0x7A) - Digital Pulls
+ */
+#define WM8993_MCLK_PU 0x0080 /* MCLK_PU */
+#define WM8993_MCLK_PU_MASK 0x0080 /* MCLK_PU */
+#define WM8993_MCLK_PU_SHIFT 7 /* MCLK_PU */
+#define WM8993_MCLK_PU_WIDTH 1 /* MCLK_PU */
+#define WM8993_MCLK_PD 0x0040 /* MCLK_PD */
+#define WM8993_MCLK_PD_MASK 0x0040 /* MCLK_PD */
+#define WM8993_MCLK_PD_SHIFT 6 /* MCLK_PD */
+#define WM8993_MCLK_PD_WIDTH 1 /* MCLK_PD */
+#define WM8993_DACDAT_PU 0x0020 /* DACDAT_PU */
+#define WM8993_DACDAT_PU_MASK 0x0020 /* DACDAT_PU */
+#define WM8993_DACDAT_PU_SHIFT 5 /* DACDAT_PU */
+#define WM8993_DACDAT_PU_WIDTH 1 /* DACDAT_PU */
+#define WM8993_DACDAT_PD 0x0010 /* DACDAT_PD */
+#define WM8993_DACDAT_PD_MASK 0x0010 /* DACDAT_PD */
+#define WM8993_DACDAT_PD_SHIFT 4 /* DACDAT_PD */
+#define WM8993_DACDAT_PD_WIDTH 1 /* DACDAT_PD */
+#define WM8993_LRCLK_PU 0x0008 /* LRCLK_PU */
+#define WM8993_LRCLK_PU_MASK 0x0008 /* LRCLK_PU */
+#define WM8993_LRCLK_PU_SHIFT 3 /* LRCLK_PU */
+#define WM8993_LRCLK_PU_WIDTH 1 /* LRCLK_PU */
+#define WM8993_LRCLK_PD 0x0004 /* LRCLK_PD */
+#define WM8993_LRCLK_PD_MASK 0x0004 /* LRCLK_PD */
+#define WM8993_LRCLK_PD_SHIFT 2 /* LRCLK_PD */
+#define WM8993_LRCLK_PD_WIDTH 1 /* LRCLK_PD */
+#define WM8993_BCLK_PU 0x0002 /* BCLK_PU */
+#define WM8993_BCLK_PU_MASK 0x0002 /* BCLK_PU */
+#define WM8993_BCLK_PU_SHIFT 1 /* BCLK_PU */
+#define WM8993_BCLK_PU_WIDTH 1 /* BCLK_PU */
+#define WM8993_BCLK_PD 0x0001 /* BCLK_PD */
+#define WM8993_BCLK_PD_MASK 0x0001 /* BCLK_PD */
+#define WM8993_BCLK_PD_SHIFT 0 /* BCLK_PD */
+#define WM8993_BCLK_PD_WIDTH 1 /* BCLK_PD */
+
+/*
+ * R123 (0x7B) - DRC Control 1
+ */
+#define WM8993_DRC_ENA 0x8000 /* DRC_ENA */
+#define WM8993_DRC_ENA_MASK 0x8000 /* DRC_ENA */
+#define WM8993_DRC_ENA_SHIFT 15 /* DRC_ENA */
+#define WM8993_DRC_ENA_WIDTH 1 /* DRC_ENA */
+#define WM8993_DRC_DAC_PATH 0x4000 /* DRC_DAC_PATH */
+#define WM8993_DRC_DAC_PATH_MASK 0x4000 /* DRC_DAC_PATH */
+#define WM8993_DRC_DAC_PATH_SHIFT 14 /* DRC_DAC_PATH */
+#define WM8993_DRC_DAC_PATH_WIDTH 1 /* DRC_DAC_PATH */
+#define WM8993_DRC_SMOOTH_ENA 0x0800 /* DRC_SMOOTH_ENA */
+#define WM8993_DRC_SMOOTH_ENA_MASK 0x0800 /* DRC_SMOOTH_ENA */
+#define WM8993_DRC_SMOOTH_ENA_SHIFT 11 /* DRC_SMOOTH_ENA */
+#define WM8993_DRC_SMOOTH_ENA_WIDTH 1 /* DRC_SMOOTH_ENA */
+#define WM8993_DRC_QR_ENA 0x0400 /* DRC_QR_ENA */
+#define WM8993_DRC_QR_ENA_MASK 0x0400 /* DRC_QR_ENA */
+#define WM8993_DRC_QR_ENA_SHIFT 10 /* DRC_QR_ENA */
+#define WM8993_DRC_QR_ENA_WIDTH 1 /* DRC_QR_ENA */
+#define WM8993_DRC_ANTICLIP_ENA 0x0200 /* DRC_ANTICLIP_ENA */
+#define WM8993_DRC_ANTICLIP_ENA_MASK 0x0200 /* DRC_ANTICLIP_ENA */
+#define WM8993_DRC_ANTICLIP_ENA_SHIFT 9 /* DRC_ANTICLIP_ENA */
+#define WM8993_DRC_ANTICLIP_ENA_WIDTH 1 /* DRC_ANTICLIP_ENA */
+#define WM8993_DRC_HYST_ENA 0x0100 /* DRC_HYST_ENA */
+#define WM8993_DRC_HYST_ENA_MASK 0x0100 /* DRC_HYST_ENA */
+#define WM8993_DRC_HYST_ENA_SHIFT 8 /* DRC_HYST_ENA */
+#define WM8993_DRC_HYST_ENA_WIDTH 1 /* DRC_HYST_ENA */
+#define WM8993_DRC_THRESH_HYST_MASK 0x0030 /* DRC_THRESH_HYST - [5:4] */
+#define WM8993_DRC_THRESH_HYST_SHIFT 4 /* DRC_THRESH_HYST - [5:4] */
+#define WM8993_DRC_THRESH_HYST_WIDTH 2 /* DRC_THRESH_HYST - [5:4] */
+#define WM8993_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */
+#define WM8993_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */
+#define WM8993_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */
+#define WM8993_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */
+#define WM8993_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */
+#define WM8993_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */
+
+/*
+ * R124 (0x7C) - DRC Control 2
+ */
+#define WM8993_DRC_ATTACK_RATE_MASK 0xF000 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8993_DRC_ATTACK_RATE_SHIFT 12 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8993_DRC_ATTACK_RATE_WIDTH 4 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8993_DRC_DECAY_RATE_MASK 0x0F00 /* DRC_DECAY_RATE - [11:8] */
+#define WM8993_DRC_DECAY_RATE_SHIFT 8 /* DRC_DECAY_RATE - [11:8] */
+#define WM8993_DRC_DECAY_RATE_WIDTH 4 /* DRC_DECAY_RATE - [11:8] */
+#define WM8993_DRC_THRESH_COMP_MASK 0x00FC /* DRC_THRESH_COMP - [7:2] */
+#define WM8993_DRC_THRESH_COMP_SHIFT 2 /* DRC_THRESH_COMP - [7:2] */
+#define WM8993_DRC_THRESH_COMP_WIDTH 6 /* DRC_THRESH_COMP - [7:2] */
+
+/*
+ * R125 (0x7D) - DRC Control 3
+ */
+#define WM8993_DRC_AMP_COMP_MASK 0xF800 /* DRC_AMP_COMP - [15:11] */
+#define WM8993_DRC_AMP_COMP_SHIFT 11 /* DRC_AMP_COMP - [15:11] */
+#define WM8993_DRC_AMP_COMP_WIDTH 5 /* DRC_AMP_COMP - [15:11] */
+#define WM8993_DRC_R0_SLOPE_COMP_MASK 0x0700 /* DRC_R0_SLOPE_COMP - [10:8] */
+#define WM8993_DRC_R0_SLOPE_COMP_SHIFT 8 /* DRC_R0_SLOPE_COMP - [10:8] */
+#define WM8993_DRC_R0_SLOPE_COMP_WIDTH 3 /* DRC_R0_SLOPE_COMP - [10:8] */
+#define WM8993_DRC_FF_DELAY 0x0080 /* DRC_FF_DELAY */
+#define WM8993_DRC_FF_DELAY_MASK 0x0080 /* DRC_FF_DELAY */
+#define WM8993_DRC_FF_DELAY_SHIFT 7 /* DRC_FF_DELAY */
+#define WM8993_DRC_FF_DELAY_WIDTH 1 /* DRC_FF_DELAY */
+#define WM8993_DRC_THRESH_QR_MASK 0x000C /* DRC_THRESH_QR - [3:2] */
+#define WM8993_DRC_THRESH_QR_SHIFT 2 /* DRC_THRESH_QR - [3:2] */
+#define WM8993_DRC_THRESH_QR_WIDTH 2 /* DRC_THRESH_QR - [3:2] */
+#define WM8993_DRC_RATE_QR_MASK 0x0003 /* DRC_RATE_QR - [1:0] */
+#define WM8993_DRC_RATE_QR_SHIFT 0 /* DRC_RATE_QR - [1:0] */
+#define WM8993_DRC_RATE_QR_WIDTH 2 /* DRC_RATE_QR - [1:0] */
+
+/*
+ * R126 (0x7E) - DRC Control 4
+ */
+#define WM8993_DRC_R1_SLOPE_COMP_MASK 0xE000 /* DRC_R1_SLOPE_COMP - [15:13] */
+#define WM8993_DRC_R1_SLOPE_COMP_SHIFT 13 /* DRC_R1_SLOPE_COMP - [15:13] */
+#define WM8993_DRC_R1_SLOPE_COMP_WIDTH 3 /* DRC_R1_SLOPE_COMP - [15:13] */
+#define WM8993_DRC_STARTUP_GAIN_MASK 0x1F00 /* DRC_STARTUP_GAIN - [12:8] */
+#define WM8993_DRC_STARTUP_GAIN_SHIFT 8 /* DRC_STARTUP_GAIN - [12:8] */
+#define WM8993_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [12:8] */
+
+#endif
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 86fc57e..c64e55a 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -165,87 +165,23 @@ struct wm9081_priv {
int master;
int fll_fref;
int fll_fout;
+ int tdm_width;
struct wm9081_retune_mobile_config *retune;
};
-static int wm9081_reg_is_volatile(int reg)
+static int wm9081_volatile_register(unsigned int reg)
{
switch (reg) {
+ case WM9081_SOFTWARE_RESET:
+ return 1;
default:
return 0;
}
}
-static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u16 *cache = codec->reg_cache;
- BUG_ON(reg > WM9081_MAX_REGISTER);
- return cache[reg];
-}
-
-static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg)
-{
- struct i2c_msg xfer[2];
- u16 data;
- int ret;
- struct i2c_client *client = codec->control_data;
-
- BUG_ON(reg > WM9081_MAX_REGISTER);
-
- /* Write register */
- xfer[0].addr = client->addr;
- xfer[0].flags = 0;
- xfer[0].len = 1;
- xfer[0].buf = &reg;
-
- /* Read data */
- xfer[1].addr = client->addr;
- xfer[1].flags = I2C_M_RD;
- xfer[1].len = 2;
- xfer[1].buf = (u8 *)&data;
-
- ret = i2c_transfer(client->adapter, xfer, 2);
- if (ret != 2) {
- dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
- return 0;
- }
-
- return (data >> 8) | ((data & 0xff) << 8);
-}
-
-static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg)
-{
- if (wm9081_reg_is_volatile(reg))
- return wm9081_read_hw(codec, reg);
- else
- return wm9081_read_reg_cache(codec, reg);
-}
-
-static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u16 *cache = codec->reg_cache;
- u8 data[3];
-
- BUG_ON(reg > WM9081_MAX_REGISTER);
-
- if (!wm9081_reg_is_volatile(reg))
- cache[reg] = value;
-
- data[0] = reg;
- data[1] = value >> 8;
- data[2] = value & 0x00ff;
-
- if (codec->hw_write(codec->control_data, data, 3) == 3)
- return 0;
- else
- return -EIO;
-}
-
static int wm9081_reset(struct snd_soc_codec *codec)
{
- return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0);
+ return snd_soc_write(codec, WM9081_SOFTWARE_RESET, 0);
}
static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0);
@@ -356,7 +292,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg;
- reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+ reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2);
if (reg & WM9081_SPK_MODE)
ucontrol->value.integer.value[0] = 1;
else
@@ -375,8 +311,8 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
- unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+ unsigned int reg_pwr = snd_soc_read(codec, WM9081_POWER_MANAGEMENT);
+ unsigned int reg2 = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2);
/* Are we changing anything? */
if (ucontrol->value.integer.value[0] ==
@@ -397,7 +333,7 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol,
reg2 &= ~WM9081_SPK_MODE;
}
- wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
+ snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
return 0;
}
@@ -456,7 +392,7 @@ static int speaker_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+ unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
@@ -468,7 +404,7 @@ static int speaker_event(struct snd_soc_dapm_widget *w,
break;
}
- wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg);
+ snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg);
return 0;
}
@@ -607,7 +543,7 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
if (ret != 0)
return ret;
- reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5);
+ reg5 = snd_soc_read(codec, WM9081_FLL_CONTROL_5);
reg5 &= ~WM9081_FLL_CLK_SRC_MASK;
switch (fll_id) {
@@ -621,44 +557,44 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
}
/* Disable CLK_SYS while we reconfigure */
- clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+ clk_sys_reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_3);
if (clk_sys_reg & WM9081_CLK_SYS_ENA)
- wm9081_write(codec, WM9081_CLOCK_CONTROL_3,
+ snd_soc_write(codec, WM9081_CLOCK_CONTROL_3,
clk_sys_reg & ~WM9081_CLK_SYS_ENA);
/* Any FLL configuration change requires that the FLL be
* disabled first. */
- reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1);
+ reg1 = snd_soc_read(codec, WM9081_FLL_CONTROL_1);
reg1 &= ~WM9081_FLL_ENA;
- wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1);
/* Apply the configuration */
if (fll_div.k)
reg1 |= WM9081_FLL_FRAC_MASK;
else
reg1 &= ~WM9081_FLL_FRAC_MASK;
- wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1);
- wm9081_write(codec, WM9081_FLL_CONTROL_2,
+ snd_soc_write(codec, WM9081_FLL_CONTROL_2,
(fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) |
(fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT));
- wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_3, fll_div.k);
- reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4);
+ reg4 = snd_soc_read(codec, WM9081_FLL_CONTROL_4);
reg4 &= ~WM9081_FLL_N_MASK;
reg4 |= fll_div.n << WM9081_FLL_N_SHIFT;
- wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_4, reg4);
reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK;
reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT;
- wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_5, reg5);
/* Enable the FLL */
- wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
+ snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
/* Then bring CLK_SYS up again if it was disabled */
if (clk_sys_reg & WM9081_CLK_SYS_ENA)
- wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg);
+ snd_soc_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg);
dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
@@ -707,6 +643,10 @@ static int configure_clock(struct snd_soc_codec *codec)
target > 3000000)
break;
}
+
+ if (i == ARRAY_SIZE(clk_sys_rates))
+ return -EINVAL;
+
} else if (wm9081->fs) {
for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
new_sysclk = clk_sys_rates[i].ratio
@@ -714,6 +654,10 @@ static int configure_clock(struct snd_soc_codec *codec)
if (new_sysclk > 3000000)
break;
}
+
+ if (i == ARRAY_SIZE(clk_sys_rates))
+ return -EINVAL;
+
} else {
new_sysclk = 12288000;
}
@@ -734,19 +678,19 @@ static int configure_clock(struct snd_soc_codec *codec)
return -EINVAL;
}
- reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_1);
if (mclkdiv)
reg |= WM9081_MCLKDIV2;
else
reg &= ~WM9081_MCLKDIV2;
- wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_CLOCK_CONTROL_1, reg);
- reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+ reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_3);
if (fll)
reg |= WM9081_CLK_SRC_SEL;
else
reg &= ~WM9081_CLK_SRC_SEL;
- wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg);
+ snd_soc_write(codec, WM9081_CLOCK_CONTROL_3, reg);
dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate);
@@ -846,76 +790,76 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_PREPARE:
/* VMID=2*40k */
- reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg &= ~WM9081_VMID_SEL_MASK;
reg |= 0x2;
- wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+ snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
/* Normal bias current */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg &= ~WM9081_STBY_BIAS_ENA;
- wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
break;
case SND_SOC_BIAS_STANDBY:
/* Initial cold start */
if (codec->bias_level == SND_SOC_BIAS_OFF) {
/* Disable LINEOUT discharge */
- reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+ reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL);
reg &= ~WM9081_LINEOUT_DISCH;
- wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+ snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg);
/* Select startup bias source */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA;
- wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
/* VMID 2*4k; Soft VMID ramp enable */
- reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg |= WM9081_VMID_RAMP | 0x6;
- wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+ snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
mdelay(100);
/* Normal bias enable & soft start off */
reg |= WM9081_BIAS_ENA;
reg &= ~WM9081_VMID_RAMP;
- wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+ snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
/* Standard bias source */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg &= ~WM9081_BIAS_SRC;
- wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
}
/* VMID 2*240k */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg &= ~WM9081_VMID_SEL_MASK;
reg |= 0x40;
- wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+ snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
/* Standby bias current on */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_STBY_BIAS_ENA;
- wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
break;
case SND_SOC_BIAS_OFF:
/* Startup bias source */
- reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_BIAS_SRC;
- wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
/* Disable VMID and biases with soft ramping */
- reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
reg |= WM9081_VMID_RAMP;
- wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+ snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
/* Actively discharge LINEOUT */
- reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+ reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL);
reg |= WM9081_LINEOUT_DISCH;
- wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+ snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg);
break;
}
@@ -929,7 +873,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
{
struct snd_soc_codec *codec = dai->codec;
struct wm9081_priv *wm9081 = codec->private_data;
- unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+ unsigned int aif2 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_2);
aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV |
WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK);
@@ -1010,7 +954,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
return -EINVAL;
}
- wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+ snd_soc_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
return 0;
}
@@ -1024,47 +968,51 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
int ret, i, best, best_val, cur_val;
unsigned int clk_ctrl2, aif1, aif2, aif3, aif4;
- clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2);
+ clk_ctrl2 = snd_soc_read(codec, WM9081_CLOCK_CONTROL_2);
clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK);
- aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+ aif1 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_1);
- aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+ aif2 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_2);
aif2 &= ~WM9081_AIF_WL_MASK;
- aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3);
+ aif3 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_3);
aif3 &= ~WM9081_BCLK_DIV_MASK;
- aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4);
+ aif4 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_4);
aif4 &= ~WM9081_LRCLK_RATE_MASK;
- /* What BCLK do we need? */
wm9081->fs = params_rate(params);
- wm9081->bclk = 2 * wm9081->fs;
- switch (params_format(params)) {
- case SNDRV_PCM_FORMAT_S16_LE:
- wm9081->bclk *= 16;
- break;
- case SNDRV_PCM_FORMAT_S20_3LE:
- wm9081->bclk *= 20;
- aif2 |= 0x4;
- break;
- case SNDRV_PCM_FORMAT_S24_LE:
- wm9081->bclk *= 24;
- aif2 |= 0x8;
- break;
- case SNDRV_PCM_FORMAT_S32_LE:
- wm9081->bclk *= 32;
- aif2 |= 0xc;
- break;
- default:
- return -EINVAL;
- }
- if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) {
+ if (wm9081->tdm_width) {
+ /* If TDM is set up then that fixes our BCLK. */
int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >>
WM9081_AIFDAC_TDM_MODE_SHIFT) + 1;
- wm9081->bclk *= slots;
+
+ wm9081->bclk = wm9081->fs * wm9081->tdm_width * slots;
+ } else {
+ /* Otherwise work out a BCLK from the sample size */
+ wm9081->bclk = 2 * wm9081->fs;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ wm9081->bclk *= 16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ wm9081->bclk *= 20;
+ aif2 |= 0x4;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ wm9081->bclk *= 24;
+ aif2 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ wm9081->bclk *= 32;
+ aif2 |= 0xc;
+ break;
+ default:
+ return -EINVAL;
+ }
}
dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk);
@@ -1149,22 +1097,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
s->name, s->rate);
/* If the EQ is enabled then disable it while we write out */
- eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA;
+ eq1 = snd_soc_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA;
if (eq1 & WM9081_EQ_ENA)
- wm9081_write(codec, WM9081_EQ_1, 0);
+ snd_soc_write(codec, WM9081_EQ_1, 0);
/* Write out the other values */
for (i = 1; i < ARRAY_SIZE(s->config); i++)
- wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]);
+ snd_soc_write(codec, WM9081_EQ_1 + i, s->config[i]);
eq1 |= (s->config[0] & ~WM9081_EQ_ENA);
- wm9081_write(codec, WM9081_EQ_1, eq1);
+ snd_soc_write(codec, WM9081_EQ_1, eq1);
}
- wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2);
- wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
- wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3);
- wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4);
+ snd_soc_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2);
+ snd_soc_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+ snd_soc_write(codec, WM9081_AUDIO_INTERFACE_3, aif3);
+ snd_soc_write(codec, WM9081_AUDIO_INTERFACE_4, aif4);
return 0;
}
@@ -1174,14 +1122,14 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
struct snd_soc_codec *codec = codec_dai->codec;
unsigned int reg;
- reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2);
+ reg = snd_soc_read(codec, WM9081_DAC_DIGITAL_2);
if (mute)
reg |= WM9081_DAC_MUTE;
else
reg &= ~WM9081_DAC_MUTE;
- wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg);
+ snd_soc_write(codec, WM9081_DAC_DIGITAL_2, reg);
return 0;
}
@@ -1207,19 +1155,25 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
}
static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
- unsigned int mask, int slots)
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
struct snd_soc_codec *codec = dai->codec;
- unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+ struct wm9081_priv *wm9081 = codec->private_data;
+ unsigned int aif1 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_1);
aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK);
- if (slots < 1 || slots > 4)
+ if (slots < 0 || slots > 4)
return -EINVAL;
+ wm9081->tdm_width = slot_width;
+
+ if (slots == 0)
+ slots = 1;
+
aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT;
- switch (mask) {
+ switch (rx_mask) {
case 1:
break;
case 2:
@@ -1235,7 +1189,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
return -EINVAL;
}
- wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1);
+ snd_soc_write(codec, WM9081_AUDIO_INTERFACE_1, aif1);
return 0;
}
@@ -1357,7 +1311,7 @@ static int wm9081_resume(struct platform_device *pdev)
if (i == WM9081_SOFTWARE_RESET)
continue;
- wm9081_write(codec, i, reg_cache[i]);
+ snd_soc_write(codec, i, reg_cache[i]);
}
wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1377,7 +1331,8 @@ struct snd_soc_codec_device soc_codec_dev_wm9081 = {
};
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081);
-static int wm9081_register(struct wm9081_priv *wm9081)
+static int wm9081_register(struct wm9081_priv *wm9081,
+ enum snd_soc_control_type control)
{
struct snd_soc_codec *codec = &wm9081->codec;
int ret;
@@ -1396,19 +1351,24 @@ static int wm9081_register(struct wm9081_priv *wm9081)
codec->private_data = wm9081;
codec->name = "WM9081";
codec->owner = THIS_MODULE;
- codec->read = wm9081_read;
- codec->write = wm9081_write;
codec->dai = &wm9081_dai;
codec->num_dai = 1;
codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache);
codec->reg_cache = &wm9081->reg_cache;
codec->bias_level = SND_SOC_BIAS_OFF;
codec->set_bias_level = wm9081_set_bias_level;
+ codec->volatile_register = wm9081_volatile_register;
memcpy(codec->reg_cache, wm9081_reg_defaults,
sizeof(wm9081_reg_defaults));
- reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET);
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET);
if (reg != 0x9081) {
dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg);
ret = -EINVAL;
@@ -1424,10 +1384,10 @@ static int wm9081_register(struct wm9081_priv *wm9081)
wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Enable zero cross by default */
- reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT);
- wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC);
- reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA);
- wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA,
+ reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT);
+ snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC);
+ reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA);
+ snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA,
reg | WM9081_SPKPGAZC);
wm9081_dai.dev = codec->dev;
@@ -1482,7 +1442,7 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
codec->dev = &i2c->dev;
- return wm9081_register(wm9081);
+ return wm9081_register(wm9081, SND_SOC_I2C);
}
static __devexit int wm9081_i2c_remove(struct i2c_client *client)
@@ -1492,6 +1452,21 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client)
return 0;
}
+#ifdef CONFIG_PM
+static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+ return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm9081_i2c_resume(struct i2c_client *client)
+{
+ return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm9081_i2c_suspend NULL
+#define wm9081_i2c_resume NULL
+#endif
+
static const struct i2c_device_id wm9081_i2c_id[] = {
{ "wm9081", 0 },
{ }
@@ -1505,6 +1480,8 @@ static struct i2c_driver wm9081_i2c_driver = {
},
.probe = wm9081_i2c_probe,
.remove = __devexit_p(wm9081_i2c_remove),
+ .suspend = wm9081_i2c_suspend,
+ .resume = wm9081_i2c_resume,
.id_table = wm9081_i2c_id,
};
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index fa88b46..e7d2840 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -406,7 +406,7 @@ static int wm9705_soc_probe(struct platform_device *pdev)
ret = snd_soc_init_card(socdev);
if (ret < 0) {
printk(KERN_ERR "wm9705: failed to register card\n");
- goto pcm_err;
+ goto reset_err;
}
return 0;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
new file mode 100644
index 0000000..e542027
--- /dev/null
+++ b/sound/soc/codecs/wm_hubs.c
@@ -0,0 +1,743 @@
+/*
+ * wm_hubs.c -- WM8993/4 common code
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8993.h"
+#include "wm_hubs.h"
+
+const DECLARE_TLV_DB_SCALE(wm_hubs_spkmix_tlv, -300, 300, 0);
+EXPORT_SYMBOL_GPL(wm_hubs_spkmix_tlv);
+
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(inmix_sw_tlv, 0, 3000, 0);
+static const DECLARE_TLV_DB_SCALE(inmix_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(earpiece_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
+static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
+static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
+static const unsigned int spkboost_tlv[] = {
+ TLV_DB_RANGE_HEAD(7),
+ 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
+ 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(line_tlv, -600, 600, 0);
+
+static const char *speaker_ref_text[] = {
+ "SPKVDD/2",
+ "VMID",
+};
+
+static const struct soc_enum speaker_ref =
+ SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text);
+
+static const char *speaker_mode_text[] = {
+ "Class D",
+ "Class AB",
+};
+
+static const struct soc_enum speaker_mode =
+ SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
+
+static void wait_for_dc_servo(struct snd_soc_codec *codec)
+{
+ unsigned int reg;
+ int count = 0;
+
+ dev_dbg(codec->dev, "Waiting for DC servo...\n");
+ do {
+ count++;
+ msleep(1);
+ reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
+ dev_dbg(codec->dev, "DC servo status: %x\n", reg);
+ } while ((reg & WM8993_DCS_CAL_COMPLETE_MASK)
+ != WM8993_DCS_CAL_COMPLETE_MASK && count < 1000);
+
+ if ((reg & WM8993_DCS_CAL_COMPLETE_MASK)
+ != WM8993_DCS_CAL_COMPLETE_MASK)
+ dev_err(codec->dev, "Timed out waiting for DC Servo\n");
+}
+
+/*
+ * Update the DC servo calibration on gain changes
+ */
+static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int ret;
+
+ ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+
+ /* Only need to do this if the outputs are active */
+ if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1)
+ & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA))
+ snd_soc_update_bits(codec,
+ WM8993_DC_SERVO_0,
+ WM8993_DCS_TRIG_SINGLE_0 |
+ WM8993_DCS_TRIG_SINGLE_1,
+ WM8993_DCS_TRIG_SINGLE_0 |
+ WM8993_DCS_TRIG_SINGLE_1);
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new analogue_snd_controls[] = {
+SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
+ inpga_tlv),
+SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+
+SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
+ inpga_tlv),
+SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+
+
+SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
+ inpga_tlv),
+SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+
+SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
+ inpga_tlv),
+SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+
+SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
+ inmix_sw_tlv),
+SOC_SINGLE_TLV("MIXINL IN1L Volume", WM8993_INPUT_MIXER3, 4, 1, 0,
+ inmix_sw_tlv),
+SOC_SINGLE_TLV("MIXINL Output Record Volume", WM8993_INPUT_MIXER3, 0, 7, 0,
+ inmix_tlv),
+SOC_SINGLE_TLV("MIXINL IN1LP Volume", WM8993_INPUT_MIXER5, 6, 7, 0, inmix_tlv),
+SOC_SINGLE_TLV("MIXINL Direct Voice Volume", WM8993_INPUT_MIXER5, 0, 6, 0,
+ inmix_tlv),
+
+SOC_SINGLE_TLV("MIXINR IN2R Volume", WM8993_INPUT_MIXER4, 7, 1, 0,
+ inmix_sw_tlv),
+SOC_SINGLE_TLV("MIXINR IN1R Volume", WM8993_INPUT_MIXER4, 4, 1, 0,
+ inmix_sw_tlv),
+SOC_SINGLE_TLV("MIXINR Output Record Volume", WM8993_INPUT_MIXER4, 0, 7, 0,
+ inmix_tlv),
+SOC_SINGLE_TLV("MIXINR IN1RP Volume", WM8993_INPUT_MIXER6, 6, 7, 0, inmix_tlv),
+SOC_SINGLE_TLV("MIXINR Direct Voice Volume", WM8993_INPUT_MIXER6, 0, 6, 0,
+ inmix_tlv),
+
+SOC_SINGLE_TLV("Left Output Mixer IN2RN Volume", WM8993_OUTPUT_MIXER5, 6, 7, 1,
+ outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer IN2LN Volume", WM8993_OUTPUT_MIXER3, 6, 7, 1,
+ outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer IN2LP Volume", WM8993_OUTPUT_MIXER3, 9, 7, 1,
+ outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer IN1L Volume", WM8993_OUTPUT_MIXER3, 0, 7, 1,
+ outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer IN1R Volume", WM8993_OUTPUT_MIXER3, 3, 7, 1,
+ outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer Right Input Volume",
+ WM8993_OUTPUT_MIXER5, 3, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer Left Input Volume",
+ WM8993_OUTPUT_MIXER5, 0, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Left Output Mixer DAC Volume", WM8993_OUTPUT_MIXER5, 9, 7, 1,
+ outmix_tlv),
+
+SOC_SINGLE_TLV("Right Output Mixer IN2LN Volume",
+ WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer IN2RN Volume",
+ WM8993_OUTPUT_MIXER4, 6, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer IN1L Volume",
+ WM8993_OUTPUT_MIXER4, 3, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer IN1R Volume",
+ WM8993_OUTPUT_MIXER4, 0, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer IN2RP Volume",
+ WM8993_OUTPUT_MIXER4, 9, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Left Input Volume",
+ WM8993_OUTPUT_MIXER6, 3, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Right Input Volume",
+ WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv),
+SOC_SINGLE_TLV("Right Output Mixer DAC Volume",
+ WM8993_OUTPUT_MIXER6, 9, 7, 1, outmix_tlv),
+
+SOC_DOUBLE_R_TLV("Output Volume", WM8993_LEFT_OPGA_VOLUME,
+ WM8993_RIGHT_OPGA_VOLUME, 0, 63, 0, outpga_tlv),
+SOC_DOUBLE_R("Output Switch", WM8993_LEFT_OPGA_VOLUME,
+ WM8993_RIGHT_OPGA_VOLUME, 6, 1, 0),
+SOC_DOUBLE_R("Output ZC Switch", WM8993_LEFT_OPGA_VOLUME,
+ WM8993_RIGHT_OPGA_VOLUME, 7, 1, 0),
+
+SOC_SINGLE("Earpiece Switch", WM8993_HPOUT2_VOLUME, 5, 1, 1),
+SOC_SINGLE_TLV("Earpiece Volume", WM8993_HPOUT2_VOLUME, 4, 1, 1, earpiece_tlv),
+
+SOC_SINGLE_TLV("SPKL Input Volume", WM8993_SPKMIXL_ATTENUATION,
+ 5, 1, 1, wm_hubs_spkmix_tlv),
+SOC_SINGLE_TLV("SPKL IN1LP Volume", WM8993_SPKMIXL_ATTENUATION,
+ 4, 1, 1, wm_hubs_spkmix_tlv),
+SOC_SINGLE_TLV("SPKL Output Volume", WM8993_SPKMIXL_ATTENUATION,
+ 3, 1, 1, wm_hubs_spkmix_tlv),
+
+SOC_SINGLE_TLV("SPKR Input Volume", WM8993_SPKMIXR_ATTENUATION,
+ 5, 1, 1, wm_hubs_spkmix_tlv),
+SOC_SINGLE_TLV("SPKR IN1RP Volume", WM8993_SPKMIXR_ATTENUATION,
+ 4, 1, 1, wm_hubs_spkmix_tlv),
+SOC_SINGLE_TLV("SPKR Output Volume", WM8993_SPKMIXR_ATTENUATION,
+ 3, 1, 1, wm_hubs_spkmix_tlv),
+
+SOC_DOUBLE_R_TLV("Speaker Mixer Volume",
+ WM8993_SPKMIXL_ATTENUATION, WM8993_SPKMIXR_ATTENUATION,
+ 0, 3, 1, spkmixout_tlv),
+SOC_DOUBLE_R_TLV("Speaker Volume",
+ WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
+ 0, 63, 0, outpga_tlv),
+SOC_DOUBLE_R("Speaker Switch",
+ WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
+ 6, 1, 0),
+SOC_DOUBLE_R("Speaker ZC Switch",
+ WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
+ 7, 1, 0),
+SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0,
+ spkboost_tlv),
+SOC_ENUM("Speaker Reference", speaker_ref),
+SOC_ENUM("Speaker Mode", speaker_mode),
+
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .tlv.p = outpga_tlv,
+ .info = snd_soc_info_volsw_2r,
+ .get = snd_soc_get_volsw_2r, .put = wm8993_put_dc_servo,
+ .private_value = (unsigned long)&(struct soc_mixer_control) {
+ .reg = WM8993_LEFT_OUTPUT_VOLUME,
+ .rreg = WM8993_RIGHT_OUTPUT_VOLUME,
+ .shift = 0, .max = 63
+ },
+},
+SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME,
+ WM8993_RIGHT_OUTPUT_VOLUME, 6, 1, 0),
+SOC_DOUBLE_R("Headphone ZC Switch", WM8993_LEFT_OUTPUT_VOLUME,
+ WM8993_RIGHT_OUTPUT_VOLUME, 7, 1, 0),
+
+SOC_SINGLE("LINEOUT1N Switch", WM8993_LINE_OUTPUTS_VOLUME, 6, 1, 1),
+SOC_SINGLE("LINEOUT1P Switch", WM8993_LINE_OUTPUTS_VOLUME, 5, 1, 1),
+SOC_SINGLE_TLV("LINEOUT1 Volume", WM8993_LINE_OUTPUTS_VOLUME, 4, 1, 1,
+ line_tlv),
+
+SOC_SINGLE("LINEOUT2N Switch", WM8993_LINE_OUTPUTS_VOLUME, 2, 1, 1),
+SOC_SINGLE("LINEOUT2P Switch", WM8993_LINE_OUTPUTS_VOLUME, 1, 1, 1),
+SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1,
+ line_tlv),
+};
+
+static int hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int reg = snd_soc_read(codec, WM8993_ANALOGUE_HP_0);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
+ WM8993_CP_ENA, WM8993_CP_ENA);
+
+ msleep(5);
+
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
+ WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA);
+
+ reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY;
+ snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
+
+ /* Start the DC servo */
+ snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
+ 0xFFFF,
+ WM8993_DCS_ENA_CHAN_0 |
+ WM8993_DCS_ENA_CHAN_1 |
+ WM8993_DCS_TRIG_STARTUP_1 |
+ WM8993_DCS_TRIG_STARTUP_0);
+ wait_for_dc_servo(codec);
+
+ reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT |
+ WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT;
+ snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ reg &= ~(WM8993_HPOUT1L_RMV_SHORT |
+ WM8993_HPOUT1L_DLY |
+ WM8993_HPOUT1L_OUTP |
+ WM8993_HPOUT1R_RMV_SHORT |
+ WM8993_HPOUT1R_DLY |
+ WM8993_HPOUT1R_OUTP);
+
+ snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
+ 0xffff, 0);
+
+ snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
+ snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+ WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
+ 0);
+
+ snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
+ WM8993_CP_ENA, 0);
+ break;
+ }
+
+ return 0;
+}
+
+static int earpiece_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *control, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 reg = snd_soc_read(codec, WM8993_ANTIPOP1) & ~WM8993_HPOUT2_IN_ENA;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ reg |= WM8993_HPOUT2_IN_ENA;
+ snd_soc_write(codec, WM8993_ANTIPOP1, reg);
+ udelay(50);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_write(codec, WM8993_ANTIPOP1, reg);
+ break;
+
+ default:
+ BUG();
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new in1l_pga[] = {
+SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new in1r_pga[] = {
+SOC_DAPM_SINGLE("IN1RP Switch", WM8993_INPUT_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1RN Switch", WM8993_INPUT_MIXER2, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new in2l_pga[] = {
+SOC_DAPM_SINGLE("IN2LP Switch", WM8993_INPUT_MIXER2, 7, 1, 0),
+SOC_DAPM_SINGLE("IN2LN Switch", WM8993_INPUT_MIXER2, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new in2r_pga[] = {
+SOC_DAPM_SINGLE("IN2RP Switch", WM8993_INPUT_MIXER2, 3, 1, 0),
+SOC_DAPM_SINGLE("IN2RN Switch", WM8993_INPUT_MIXER2, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new mixinl[] = {
+SOC_DAPM_SINGLE("IN2L Switch", WM8993_INPUT_MIXER3, 8, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_INPUT_MIXER3, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new mixinr[] = {
+SOC_DAPM_SINGLE("IN2R Switch", WM8993_INPUT_MIXER4, 8, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_output_mixer[] = {
+SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
+SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
+SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
+SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
+SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
+SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_mixer[] = {
+SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
+SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new earpiece_mixer[] = {
+SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_HPOUT2_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_HPOUT2_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_HPOUT2_MIXER, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_speaker_boost[] = {
+SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 5, 1, 0),
+SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 4, 1, 0),
+SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_speaker_boost[] = {
+SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 2, 1, 0),
+SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 1, 1, 0),
+SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new line1_mix[] = {
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER1, 2, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER1, 1, 1, 0),
+SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new line1n_mix[] = {
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 6, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER1, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new line1p_mix[] = {
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new line2_mix[] = {
+SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new line2n_mix[] = {
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new line2p_mix[] = {
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1LN"),
+SND_SOC_DAPM_INPUT("IN1LP"),
+SND_SOC_DAPM_INPUT("IN2LN"),
+SND_SOC_DAPM_INPUT("IN2LP/VXRN"),
+SND_SOC_DAPM_INPUT("IN1RN"),
+SND_SOC_DAPM_INPUT("IN1RP"),
+SND_SOC_DAPM_INPUT("IN2RN"),
+SND_SOC_DAPM_INPUT("IN2RP/VXRP"),
+
+SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0),
+SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0),
+
+SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
+ in1l_pga, ARRAY_SIZE(in1l_pga)),
+SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
+ in1r_pga, ARRAY_SIZE(in1r_pga)),
+
+SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0,
+ in2l_pga, ARRAY_SIZE(in2l_pga)),
+SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0,
+ in2r_pga, ARRAY_SIZE(in2r_pga)),
+
+/* Dummy widgets to represent differential paths */
+SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0,
+ mixinl, ARRAY_SIZE(mixinl)),
+SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0,
+ mixinr, ARRAY_SIZE(mixinr)),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8993_POWER_MANAGEMENT_3, 5, 0,
+ left_output_mixer, ARRAY_SIZE(left_output_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0,
+ right_output_mixer, ARRAY_SIZE(right_output_mixer)),
+
+SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0,
+ NULL, 0,
+ hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
+ earpiece_mixer, ARRAY_SIZE(earpiece_mixer)),
+SND_SOC_DAPM_PGA_E("Earpiece Driver", WM8993_POWER_MANAGEMENT_1, 11, 0,
+ NULL, 0, earpiece_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0,
+ left_speaker_boost, ARRAY_SIZE(left_speaker_boost)),
+SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0,
+ right_speaker_boost, ARRAY_SIZE(right_speaker_boost)),
+
+SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+ line1_mix, ARRAY_SIZE(line1_mix)),
+SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+ line2_mix, ARRAY_SIZE(line2_mix)),
+
+SND_SOC_DAPM_MIXER("LINEOUT1N Mixer", SND_SOC_NOPM, 0, 0,
+ line1n_mix, ARRAY_SIZE(line1n_mix)),
+SND_SOC_DAPM_MIXER("LINEOUT1P Mixer", SND_SOC_NOPM, 0, 0,
+ line1p_mix, ARRAY_SIZE(line1p_mix)),
+SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0,
+ line2n_mix, ARRAY_SIZE(line2n_mix)),
+SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0,
+ line2p_mix, ARRAY_SIZE(line2p_mix)),
+
+SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2P"),
+SND_SOC_DAPM_OUTPUT("HPOUT2N"),
+SND_SOC_DAPM_OUTPUT("LINEOUT1P"),
+SND_SOC_DAPM_OUTPUT("LINEOUT1N"),
+SND_SOC_DAPM_OUTPUT("LINEOUT2P"),
+SND_SOC_DAPM_OUTPUT("LINEOUT2N"),
+};
+
+static const struct snd_soc_dapm_route analogue_routes[] = {
+ { "IN1L PGA", "IN1LP Switch", "IN1LP" },
+ { "IN1L PGA", "IN1LN Switch", "IN1LN" },
+
+ { "IN1R PGA", "IN1RP Switch", "IN1RP" },
+ { "IN1R PGA", "IN1RN Switch", "IN1RN" },
+
+ { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" },
+ { "IN2L PGA", "IN2LN Switch", "IN2LN" },
+
+ { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" },
+ { "IN2R PGA", "IN2RN Switch", "IN2RN" },
+
+ { "Direct Voice", NULL, "IN2LP/VXRN" },
+ { "Direct Voice", NULL, "IN2RP/VXRP" },
+
+ { "MIXINL", "IN1L Switch", "IN1L PGA" },
+ { "MIXINL", "IN2L Switch", "IN2L PGA" },
+ { "MIXINL", NULL, "Direct Voice" },
+ { "MIXINL", NULL, "IN1LP" },
+ { "MIXINL", NULL, "Left Output Mixer" },
+
+ { "MIXINR", "IN1R Switch", "IN1R PGA" },
+ { "MIXINR", "IN2R Switch", "IN2R PGA" },
+ { "MIXINR", NULL, "Direct Voice" },
+ { "MIXINR", NULL, "IN1RP" },
+ { "MIXINR", NULL, "Right Output Mixer" },
+
+ { "ADCL", NULL, "MIXINL" },
+ { "ADCR", NULL, "MIXINR" },
+
+ { "Left Output Mixer", "Left Input Switch", "MIXINL" },
+ { "Left Output Mixer", "Right Input Switch", "MIXINR" },
+ { "Left Output Mixer", "IN2RN Switch", "IN2RN" },
+ { "Left Output Mixer", "IN2LN Switch", "IN2LN" },
+ { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" },
+ { "Left Output Mixer", "IN1L Switch", "IN1L PGA" },
+ { "Left Output Mixer", "IN1R Switch", "IN1R PGA" },
+
+ { "Right Output Mixer", "Left Input Switch", "MIXINL" },
+ { "Right Output Mixer", "Right Input Switch", "MIXINR" },
+ { "Right Output Mixer", "IN2LN Switch", "IN2LN" },
+ { "Right Output Mixer", "IN2RN Switch", "IN2RN" },
+ { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" },
+ { "Right Output Mixer", "IN1L Switch", "IN1L PGA" },
+ { "Right Output Mixer", "IN1R Switch", "IN1R PGA" },
+
+ { "Left Output PGA", NULL, "Left Output Mixer" },
+ { "Left Output PGA", NULL, "TOCLK" },
+
+ { "Right Output PGA", NULL, "Right Output Mixer" },
+ { "Right Output PGA", NULL, "TOCLK" },
+
+ { "Earpiece Mixer", "Direct Voice Switch", "Direct Voice" },
+ { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" },
+ { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" },
+
+ { "Earpiece Driver", NULL, "Earpiece Mixer" },
+ { "HPOUT2N", NULL, "Earpiece Driver" },
+ { "HPOUT2P", NULL, "Earpiece Driver" },
+
+ { "SPKL", "Input Switch", "MIXINL" },
+ { "SPKL", "IN1LP Switch", "IN1LP" },
+ { "SPKL", "Output Switch", "Left Output Mixer" },
+ { "SPKL", NULL, "TOCLK" },
+
+ { "SPKR", "Input Switch", "MIXINR" },
+ { "SPKR", "IN1RP Switch", "IN1RP" },
+ { "SPKR", "Output Switch", "Right Output Mixer" },
+ { "SPKR", NULL, "TOCLK" },
+
+ { "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
+ { "SPKL Boost", "SPKL Switch", "SPKL" },
+ { "SPKL Boost", "SPKR Switch", "SPKR" },
+
+ { "SPKR Boost", "Direct Voice Switch", "Direct Voice" },
+ { "SPKR Boost", "SPKR Switch", "SPKR" },
+ { "SPKR Boost", "SPKL Switch", "SPKL" },
+
+ { "SPKL Driver", NULL, "SPKL Boost" },
+ { "SPKL Driver", NULL, "CLK_SYS" },
+
+ { "SPKR Driver", NULL, "SPKR Boost" },
+ { "SPKR Driver", NULL, "CLK_SYS" },
+
+ { "SPKOUTLP", NULL, "SPKL Driver" },
+ { "SPKOUTLN", NULL, "SPKL Driver" },
+ { "SPKOUTRP", NULL, "SPKR Driver" },
+ { "SPKOUTRN", NULL, "SPKR Driver" },
+
+ { "Left Headphone Mux", "Mixer", "Left Output Mixer" },
+ { "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+
+ { "Headphone PGA", NULL, "Left Headphone Mux" },
+ { "Headphone PGA", NULL, "Right Headphone Mux" },
+ { "Headphone PGA", NULL, "CLK_SYS" },
+
+ { "HPOUT1L", NULL, "Headphone PGA" },
+ { "HPOUT1R", NULL, "Headphone PGA" },
+
+ { "LINEOUT1N", NULL, "LINEOUT1N Driver" },
+ { "LINEOUT1P", NULL, "LINEOUT1P Driver" },
+ { "LINEOUT2N", NULL, "LINEOUT2N Driver" },
+ { "LINEOUT2P", NULL, "LINEOUT2P Driver" },
+};
+
+static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
+ { "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" },
+ { "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" },
+ { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" },
+
+ { "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" },
+ { "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" },
+};
+
+static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" },
+ { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" },
+
+ { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" },
+
+ { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
+ { "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" },
+};
+
+static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
+ { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
+ { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
+ { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" },
+
+ { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
+ { "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" },
+};
+
+static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" },
+ { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" },
+
+ { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" },
+
+ { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
+ { "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" },
+};
+
+int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
+{
+ /* Latch volume update bits & default ZC on */
+ snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8993_IN1_VU, WM8993_IN1_VU);
+ snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8993_IN1_VU, WM8993_IN1_VU);
+ snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8993_IN2_VU, WM8993_IN2_VU);
+ snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8993_IN2_VU, WM8993_IN2_VU);
+
+ snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT,
+ WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
+
+ snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME,
+ WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC);
+ snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME,
+ WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC,
+ WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC);
+
+ snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME,
+ WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC);
+ snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME,
+ WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU,
+ WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU);
+
+ snd_soc_add_controls(codec, analogue_snd_controls,
+ ARRAY_SIZE(analogue_snd_controls));
+
+ snd_soc_dapm_new_controls(codec, analogue_dapm_widgets,
+ ARRAY_SIZE(analogue_dapm_widgets));
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls);
+
+int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
+ int lineout1_diff, int lineout2_diff)
+{
+ snd_soc_dapm_add_routes(codec, analogue_routes,
+ ARRAY_SIZE(analogue_routes));
+
+ if (lineout1_diff)
+ snd_soc_dapm_add_routes(codec,
+ lineout1_diff_routes,
+ ARRAY_SIZE(lineout1_diff_routes));
+ else
+ snd_soc_dapm_add_routes(codec,
+ lineout1_se_routes,
+ ARRAY_SIZE(lineout1_se_routes));
+
+ if (lineout2_diff)
+ snd_soc_dapm_add_routes(codec,
+ lineout2_diff_routes,
+ ARRAY_SIZE(lineout2_diff_routes));
+ else
+ snd_soc_dapm_add_routes(codec,
+ lineout2_se_routes,
+ ARRAY_SIZE(lineout2_se_routes));
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes);
+
+MODULE_DESCRIPTION("Shared support for Wolfson hubs products");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
new file mode 100644
index 0000000..ec09cb6
--- /dev/null
+++ b/sound/soc/codecs/wm_hubs.h
@@ -0,0 +1,24 @@
+/*
+ * wm_hubs.h -- WM899x common code
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM_HUBS_H
+#define _WM_HUBS_H
+
+struct snd_soc_codec;
+
+extern const unsigned int wm_hubs_spkmix_tlv[];
+
+extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
+extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int);
+
+#endif
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 411a710..4dfd4ad 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -9,6 +9,9 @@ config SND_DAVINCI_SOC
config SND_DAVINCI_SOC_I2S
tristate
+config SND_DAVINCI_SOC_MCASP
+ tristate
+
config SND_DAVINCI_SOC_EVM
tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
depends on SND_DAVINCI_SOC
@@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM
Say Y if you want to add support for SoC audio on TI
DaVinci DM6446 or DM355 EVM platforms.
+config SND_DM6467_SOC_EVM
+ tristate "SoC Audio support for DaVinci DM6467 EVM"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM
+ select SND_DAVINCI_SOC_MCASP
+ select SND_SOC_TLV320AIC3X
+ select SND_SOC_SPDIF
+
+ help
+ Say Y if you want to add support for SoC audio on TI
+
config SND_DAVINCI_SOC_SFFSDR
tristate "SoC Audio support for SFFSDR"
depends on SND_DAVINCI_SOC && MACH_SFFSDR
@@ -28,3 +41,23 @@ config SND_DAVINCI_SOC_SFFSDR
help
Say Y if you want to add support for SoC audio on
Lyrtech SFFSDR board.
+
+config SND_DA830_SOC_EVM
+ tristate "SoC Audio support for DA830/OMAP-L137 EVM"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM
+ select SND_DAVINCI_SOC_MCASP
+ select SND_SOC_TLV320AIC3X
+
+ help
+ Say Y if you want to add support for SoC audio on TI
+ DA830/OMAP-L137 EVM
+
+config SND_DA850_SOC_EVM
+ tristate "SoC Audio support for DA850/OMAP-L138 EVM"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM
+ select SND_DAVINCI_SOC_MCASP
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on TI
+ DA850/OMAP-L138 EVM
+
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index ca8bae1..a6939d7 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -1,13 +1,18 @@
# DAVINCI Platform Support
snd-soc-davinci-objs := davinci-pcm.o
snd-soc-davinci-i2s-objs := davinci-i2s.o
+snd-soc-davinci-mcasp-objs:= davinci-mcasp.o
obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
+obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
# DAVINCI Machine Support
snd-soc-evm-objs := davinci-evm.o
snd-soc-sffsdr-objs := davinci-sffsdr.o
obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o
obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 58fd1cb..67414f6 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -14,6 +14,7 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -27,9 +28,10 @@
#include <mach/mux.h>
#include "../codecs/tlv320aic3x.h"
+#include "../codecs/spdif_transciever.h"
#include "davinci-pcm.h"
#include "davinci-i2s.h"
-
+#include "davinci-mcasp.h"
#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
@@ -43,7 +45,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
unsigned sysclk;
/* ASP1 on DM355 EVM is clocked by an external oscillator */
- if (machine_is_davinci_dm355_evm())
+ if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm())
sysclk = 27000000;
/* ASP0 in DM6446 EVM is clocked by U55, as configured by
@@ -53,6 +55,10 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
else if (machine_is_davinci_evm())
sysclk = 12288000;
+ else if (machine_is_davinci_da830_evm() ||
+ machine_is_davinci_da850_evm())
+ sysclk = 24576000;
+
else
return -EINVAL;
@@ -144,6 +150,32 @@ static struct snd_soc_dai_link evm_dai = {
.ops = &evm_ops,
};
+static struct snd_soc_dai_link dm6467_evm_dai[] = {
+ {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI],
+ .codec_dai = &aic3x_dai,
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+ },
+ {
+ .name = "McASP",
+ .stream_name = "spdif",
+ .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI],
+ .codec_dai = &dit_stub_dai,
+ .ops = &evm_ops,
+ },
+};
+static struct snd_soc_dai_link da8xx_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI],
+ .codec_dai = &aic3x_dai,
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
/* davinci-evm audio machine driver */
static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
@@ -152,73 +184,80 @@ static struct snd_soc_card snd_soc_card_evm = {
.num_links = 1,
};
-/* evm audio private data */
-static struct aic3x_setup_data evm_aic3x_setup = {
- .i2c_bus = 1,
- .i2c_address = 0x1b,
+/* davinci dm6467 evm audio machine driver */
+static struct snd_soc_card dm6467_snd_soc_card_evm = {
+ .name = "DaVinci DM6467 EVM",
+ .platform = &davinci_soc_platform,
+ .dai_link = dm6467_evm_dai,
+ .num_links = ARRAY_SIZE(dm6467_evm_dai),
};
+static struct snd_soc_card da830_snd_soc_card = {
+ .name = "DA830/OMAP-L137 EVM",
+ .dai_link = &da8xx_evm_dai,
+ .platform = &davinci_soc_platform,
+ .num_links = 1,
+};
+
+static struct snd_soc_card da850_snd_soc_card = {
+ .name = "DA850/OMAP-L138 EVM",
+ .dai_link = &da8xx_evm_dai,
+ .platform = &davinci_soc_platform,
+ .num_links = 1,
+};
+
+static struct aic3x_setup_data aic3x_setup;
+
/* evm audio subsystem */
static struct snd_soc_device evm_snd_devdata = {
.card = &snd_soc_card_evm,
.codec_dev = &soc_codec_dev_aic3x,
- .codec_data = &evm_aic3x_setup,
-};
-
-/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
-static struct resource evm_snd_resources[] = {
- {
- .start = DAVINCI_ASP0_BASE,
- .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
- .flags = IORESOURCE_MEM,
- },
+ .codec_data = &aic3x_setup,
};
-static struct evm_snd_platform_data evm_snd_data = {
- .tx_dma_ch = DAVINCI_DMA_ASP0_TX,
- .rx_dma_ch = DAVINCI_DMA_ASP0_RX,
+/* evm audio subsystem */
+static struct snd_soc_device dm6467_evm_snd_devdata = {
+ .card = &dm6467_snd_soc_card_evm,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &aic3x_setup,
};
-/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
-static struct resource dm335evm_snd_resources[] = {
- {
- .start = DAVINCI_ASP1_BASE,
- .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
- .flags = IORESOURCE_MEM,
- },
+/* evm audio subsystem */
+static struct snd_soc_device da830_evm_snd_devdata = {
+ .card = &da830_snd_soc_card,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &aic3x_setup,
};
-static struct evm_snd_platform_data dm335evm_snd_data = {
- .tx_dma_ch = DAVINCI_DMA_ASP1_TX,
- .rx_dma_ch = DAVINCI_DMA_ASP1_RX,
+static struct snd_soc_device da850_evm_snd_devdata = {
+ .card = &da850_snd_soc_card,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &aic3x_setup,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
- struct resource *resources;
- unsigned num_resources;
- struct evm_snd_platform_data *data;
+ struct snd_soc_device *evm_snd_dev_data;
int index;
int ret;
if (machine_is_davinci_evm()) {
- davinci_cfg_reg(DM644X_MCBSP);
-
- resources = evm_snd_resources;
- num_resources = ARRAY_SIZE(evm_snd_resources);
- data = &evm_snd_data;
+ evm_snd_dev_data = &evm_snd_devdata;
index = 0;
} else if (machine_is_davinci_dm355_evm()) {
- /* we don't use ASP1 IRQs, or we'd need to mux them ... */
- davinci_cfg_reg(DM355_EVT8_ASP1_TX);
- davinci_cfg_reg(DM355_EVT9_ASP1_RX);
-
- resources = dm335evm_snd_resources;
- num_resources = ARRAY_SIZE(dm335evm_snd_resources);
- data = &dm335evm_snd_data;
+ evm_snd_dev_data = &evm_snd_devdata;
+ index = 1;
+ } else if (machine_is_davinci_dm6467_evm()) {
+ evm_snd_dev_data = &dm6467_evm_snd_devdata;
+ index = 0;
+ } else if (machine_is_davinci_da830_evm()) {
+ evm_snd_dev_data = &da830_evm_snd_devdata;
index = 1;
+ } else if (machine_is_davinci_da850_evm()) {
+ evm_snd_dev_data = &da850_evm_snd_devdata;
+ index = 0;
} else
return -EINVAL;
@@ -226,17 +265,8 @@ static int __init evm_init(void)
if (!evm_snd_device)
return -ENOMEM;
- platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
- evm_snd_devdata.dev = &evm_snd_device->dev;
- platform_device_add_data(evm_snd_device, data, sizeof(*data));
-
- ret = platform_device_add_resources(evm_snd_device, resources,
- num_resources);
- if (ret) {
- platform_device_put(evm_snd_device);
- return ret;
- }
-
+ platform_set_drvdata(evm_snd_device, evm_snd_dev_data);
+ evm_snd_dev_data->dev = &evm_snd_device->dev;
ret = platform_device_add(evm_snd_device);
if (ret)
platform_device_put(evm_snd_device);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index b1ea52f..12a6c54 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -22,6 +22,8 @@
#include <sound/initval.h>
#include <sound/soc.h>
+#include <mach/asp.h>
+
#include "davinci-pcm.h"
@@ -63,6 +65,7 @@
#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5)
#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8)
#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16)
+#define DAVINCI_MCBSP_RCR_RFIG (1 << 18)
#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21)
#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5)
@@ -85,14 +88,6 @@
#define DAVINCI_MCBSP_PCR_FSRM (1 << 10)
#define DAVINCI_MCBSP_PCR_FSXM (1 << 11)
-#define MOD_REG_BIT(val, mask, set) do { \
- if (set) { \
- val |= mask; \
- } else { \
- val &= ~mask; \
- } \
-} while (0)
-
enum {
DAVINCI_MCBSP_WORD_8 = 0,
DAVINCI_MCBSP_WORD_12,
@@ -112,6 +107,10 @@ static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
struct davinci_mcbsp_dev {
void __iomem *base;
+#define MOD_DSP_A 0
+#define MOD_DSP_B 1
+ int mode;
+ u32 pcr;
struct clk *clk;
struct davinci_pcm_dma_params *dma_params[2];
};
@@ -127,96 +126,100 @@ static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg)
return __raw_readl(dev->base + reg);
}
-static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
+static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback)
+{
+ u32 m = playback ? DAVINCI_MCBSP_PCR_CLKXP : DAVINCI_MCBSP_PCR_CLKRP;
+ /* The clock needs to toggle to complete reset.
+ * So, fake it by toggling the clk polarity.
+ */
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr ^ m);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr);
+}
+
+static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev,
+ struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_platform *platform = socdev->card->platform;
- u32 w;
- int ret;
-
- /* Start the sample generator and enable transmitter/receiver */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ u32 spcr;
+ u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ if (spcr & mask) {
+ /* start off disabled */
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
+ spcr & ~mask);
+ toggle_clock(dev, playback);
+ }
+ if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM |
+ DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) {
+ /* Start the sample generator */
+ spcr |= DAVINCI_MCBSP_SPCR_GRST;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+ }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (playback) {
/* Stop the DMA to avoid data loss */
/* while the transmitter is out of reset to handle XSYNCERR */
if (platform->pcm_ops->trigger) {
- ret = platform->pcm_ops->trigger(substream,
+ int ret = platform->pcm_ops->trigger(substream,
SNDRV_PCM_TRIGGER_STOP);
if (ret < 0)
printk(KERN_DEBUG "Playback DMA stop failed\n");
}
/* Enable the transmitter */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ spcr |= DAVINCI_MCBSP_SPCR_XRST;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
/* wait for any unexpected frame sync error to occur */
udelay(100);
/* Disable the transmitter to clear any outstanding XSYNCERR */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ spcr &= ~DAVINCI_MCBSP_SPCR_XRST;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+ toggle_clock(dev, playback);
/* Restart the DMA */
if (platform->pcm_ops->trigger) {
- ret = platform->pcm_ops->trigger(substream,
+ int ret = platform->pcm_ops->trigger(substream,
SNDRV_PCM_TRIGGER_START);
if (ret < 0)
printk(KERN_DEBUG "Playback DMA start failed\n");
}
- /* Enable the transmitter */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
-
- } else {
-
- /* Enable the reciever */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
}
+ /* Enable transmitter or receiver */
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ spcr |= mask;
- /* Start frame sync */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM)) {
+ /* Start frame sync */
+ spcr |= DAVINCI_MCBSP_SPCR_FRST;
+ }
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
}
-static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
+static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
- u32 w;
+ u32 spcr;
/* Reset transmitter/receiver and sample rate/frame sync generators */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST |
- DAVINCI_MCBSP_SPCR_FRST, 0);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
- else
- MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ spcr &= ~(DAVINCI_MCBSP_SPCR_GRST | DAVINCI_MCBSP_SPCR_FRST);
+ spcr &= playback ? ~DAVINCI_MCBSP_SPCR_XRST : ~DAVINCI_MCBSP_SPCR_RRST;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+ toggle_clock(dev, playback);
}
static int davinci_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
+ struct snd_soc_dai *cpu_dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
-
+ struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
cpu_dai->dma_data = dev->dma_params[substream->stream];
-
return 0;
}
@@ -228,12 +231,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
unsigned int pcr;
unsigned int srgr;
- unsigned int rcr;
- unsigned int xcr;
srgr = DAVINCI_MCBSP_SRGR_FSGM |
DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
+ /* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
/* cpu is master */
@@ -258,11 +260,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
return -EINVAL;
}
- rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1);
- xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1);
+ /* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_DSP_B:
- break;
case SND_SOC_DAIFMT_I2S:
/* Davinci doesn't support TRUE I2S, but some codecs will have
* the left and right channels contiguous. This allows
@@ -282,8 +281,10 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
*/
fmt ^= SND_SOC_DAIFMT_NB_IF;
case SND_SOC_DAIFMT_DSP_A:
- rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
- xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+ dev->mode = MOD_DSP_A;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ dev->mode = MOD_DSP_B;
break;
default:
printk(KERN_ERR "%s:bad format\n", __func__);
@@ -343,9 +344,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
return -EINVAL;
}
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+ dev->pcr = pcr;
davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
return 0;
}
@@ -353,31 +353,40 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
- struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ struct davinci_pcm_dma_params *dma_params = dai->dma_data;
+ struct davinci_mcbsp_dev *dev = dai->private_data;
struct snd_interval *i = NULL;
int mcbsp_word_length;
- u32 w;
+ unsigned int rcr, xcr, srgr;
+ u32 spcr;
/* general line settings */
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
} else {
- w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+ spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
}
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
- w = DAVINCI_MCBSP_SRGR_FSGM;
- MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
+ srgr = DAVINCI_MCBSP_SRGR_FSGM;
+ srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
- MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
+ srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+ rcr = DAVINCI_MCBSP_RCR_RFIG;
+ xcr = DAVINCI_MCBSP_XCR_XFIG;
+ if (dev->mode == MOD_DSP_B) {
+ rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0);
+ xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0);
+ } else {
+ rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
+ xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+ }
/* Determine xfer data type */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
@@ -397,18 +406,31 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+ dma_params->acnt = dma_params->data_type;
+ rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1);
+ xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1);
- } else {
- w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
- MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
- DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
- davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+ rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length);
+ xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
+ else
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+ return 0;
+}
+
+static int davinci_i2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_mcbsp_dev *dev = dai->private_data;
+ int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ davinci_mcbsp_stop(dev, playback);
+ if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) {
+ /* codec is master */
+ davinci_mcbsp_start(dev, substream);
}
return 0;
}
@@ -416,35 +438,72 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
+ struct davinci_mcbsp_dev *dev = dai->private_data;
int ret = 0;
+ int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0)
+ return 0; /* return if codec is master */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- davinci_mcbsp_start(substream);
+ davinci_mcbsp_start(dev, substream);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- davinci_mcbsp_stop(substream);
+ davinci_mcbsp_stop(dev, playback);
break;
default:
ret = -EINVAL;
}
-
return ret;
}
-static int davinci_i2s_probe(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct davinci_mcbsp_dev *dev = dai->private_data;
+ int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ davinci_mcbsp_stop(dev, playback);
+}
+
+#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
+
+static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+ .startup = davinci_i2s_startup,
+ .shutdown = davinci_i2s_shutdown,
+ .prepare = davinci_i2s_prepare,
+ .trigger = davinci_i2s_trigger,
+ .hw_params = davinci_i2s_hw_params,
+ .set_fmt = davinci_i2s_set_dai_fmt,
+
+};
+
+struct snd_soc_dai davinci_i2s_dai = {
+ .name = "davinci-i2s",
+ .id = 0,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &davinci_i2s_dai_ops,
+
+};
+EXPORT_SYMBOL_GPL(davinci_i2s_dai);
+
+static int davinci_i2s_probe(struct platform_device *pdev)
{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
+ struct snd_platform_data *pdata = pdev->dev.platform_data;
struct davinci_mcbsp_dev *dev;
- struct resource *mem, *ioarea;
- struct evm_snd_platform_data *pdata;
+ struct resource *mem, *ioarea, *res;
int ret;
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -466,8 +525,6 @@ static int davinci_i2s_probe(struct platform_device *pdev,
goto err_release_region;
}
- cpu_dai->private_data = dev;
-
dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk)) {
ret = -ENODEV;
@@ -476,18 +533,37 @@ static int davinci_i2s_probe(struct platform_device *pdev,
clk_enable(dev->clk);
dev->base = (void __iomem *)IO_ADDRESS(mem->start);
- pdata = pdev->dev.platform_data;
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch;
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch;
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+ /* first TX, then RX */
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!res) {
+ dev_err(&pdev->dev, "no DMA resource\n");
+ ret = -ENXIO;
+ goto err_free_mem;
+ }
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!res) {
+ dev_err(&pdev->dev, "no DMA resource\n");
+ ret = -ENXIO;
+ goto err_free_mem;
+ }
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+
+ davinci_i2s_dai.private_data = dev;
+ ret = snd_soc_register_dai(&davinci_i2s_dai);
+ if (ret != 0)
+ goto err_free_mem;
+
return 0;
err_free_mem:
@@ -498,62 +574,40 @@ err_release_region:
return ret;
}
-static void davinci_i2s_remove(struct platform_device *pdev,
- struct snd_soc_dai *dai)
+static int davinci_i2s_remove(struct platform_device *pdev)
{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_card *card = socdev->card;
- struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
- struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+ struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data;
struct resource *mem;
+ snd_soc_unregister_dai(&davinci_i2s_dai);
clk_disable(dev->clk);
clk_put(dev->clk);
dev->clk = NULL;
-
kfree(dev);
-
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
release_mem_region(mem->start, (mem->end - mem->start) + 1);
-}
-#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
-
-static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
- .startup = davinci_i2s_startup,
- .trigger = davinci_i2s_trigger,
- .hw_params = davinci_i2s_hw_params,
- .set_fmt = davinci_i2s_set_dai_fmt,
-};
+ return 0;
+}
-struct snd_soc_dai davinci_i2s_dai = {
- .name = "davinci-i2s",
- .id = 0,
- .probe = davinci_i2s_probe,
- .remove = davinci_i2s_remove,
- .playback = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = DAVINCI_I2S_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .capture = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = DAVINCI_I2S_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .ops = &davinci_i2s_dai_ops,
+static struct platform_driver davinci_mcbsp_driver = {
+ .probe = davinci_i2s_probe,
+ .remove = davinci_i2s_remove,
+ .driver = {
+ .name = "davinci-asp",
+ .owner = THIS_MODULE,
+ },
};
-EXPORT_SYMBOL_GPL(davinci_i2s_dai);
static int __init davinci_i2s_init(void)
{
- return snd_soc_register_dai(&davinci_i2s_dai);
+ return platform_driver_register(&davinci_mcbsp_driver);
}
module_init(davinci_i2s_init);
static void __exit davinci_i2s_exit(void)
{
- snd_soc_unregister_dai(&davinci_i2s_dai);
+ platform_driver_unregister(&davinci_mcbsp_driver);
}
module_exit(davinci_i2s_exit);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
new file mode 100644
index 0000000..eca22d7
--- /dev/null
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -0,0 +1,973 @@
+/*
+ * ALSA SoC McASP Audio Layer for TI DAVINCI processor
+ *
+ * Multi-channel Audio Serial Port Driver
+ *
+ * Author: Nirmal Pandey <n-pandey@ti.com>,
+ * Suresh Rajashekara <suresh.r@ti.com>
+ * Steve Chen <schen@.mvista.com>
+ *
+ * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright: (C) 2009 Texas Instruments, India
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "davinci-pcm.h"
+#include "davinci-mcasp.h"
+
+/*
+ * McASP register definitions
+ */
+#define DAVINCI_MCASP_PID_REG 0x00
+#define DAVINCI_MCASP_PWREMUMGT_REG 0x04
+
+#define DAVINCI_MCASP_PFUNC_REG 0x10
+#define DAVINCI_MCASP_PDIR_REG 0x14
+#define DAVINCI_MCASP_PDOUT_REG 0x18
+#define DAVINCI_MCASP_PDSET_REG 0x1c
+
+#define DAVINCI_MCASP_PDCLR_REG 0x20
+
+#define DAVINCI_MCASP_TLGC_REG 0x30
+#define DAVINCI_MCASP_TLMR_REG 0x34
+
+#define DAVINCI_MCASP_GBLCTL_REG 0x44
+#define DAVINCI_MCASP_AMUTE_REG 0x48
+#define DAVINCI_MCASP_LBCTL_REG 0x4c
+
+#define DAVINCI_MCASP_TXDITCTL_REG 0x50
+
+#define DAVINCI_MCASP_GBLCTLR_REG 0x60
+#define DAVINCI_MCASP_RXMASK_REG 0x64
+#define DAVINCI_MCASP_RXFMT_REG 0x68
+#define DAVINCI_MCASP_RXFMCTL_REG 0x6c
+
+#define DAVINCI_MCASP_ACLKRCTL_REG 0x70
+#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74
+#define DAVINCI_MCASP_RXTDM_REG 0x78
+#define DAVINCI_MCASP_EVTCTLR_REG 0x7c
+
+#define DAVINCI_MCASP_RXSTAT_REG 0x80
+#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84
+#define DAVINCI_MCASP_RXCLKCHK_REG 0x88
+#define DAVINCI_MCASP_REVTCTL_REG 0x8c
+
+#define DAVINCI_MCASP_GBLCTLX_REG 0xa0
+#define DAVINCI_MCASP_TXMASK_REG 0xa4
+#define DAVINCI_MCASP_TXFMT_REG 0xa8
+#define DAVINCI_MCASP_TXFMCTL_REG 0xac
+
+#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0
+#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4
+#define DAVINCI_MCASP_TXTDM_REG 0xb8
+#define DAVINCI_MCASP_EVTCTLX_REG 0xbc
+
+#define DAVINCI_MCASP_TXSTAT_REG 0xc0
+#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4
+#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8
+#define DAVINCI_MCASP_XEVTCTL_REG 0xcc
+
+/* Left(even TDM Slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRA_REG 0x100
+/* Right(odd TDM slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRB_REG 0x118
+/* Left(even TDM slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRA_REG 0x130
+/* Right(odd TDM Slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRB_REG 0x148
+
+/* Serializer n Control Register */
+#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180
+#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \
+ (n << 2))
+
+/* Transmit Buffer for Serializer n */
+#define DAVINCI_MCASP_TXBUF_REG 0x200
+/* Receive Buffer for Serializer n */
+#define DAVINCI_MCASP_RXBUF_REG 0x280
+
+/* McASP FIFO Registers */
+#define DAVINCI_MCASP_WFIFOCTL (0x1010)
+#define DAVINCI_MCASP_WFIFOSTS (0x1014)
+#define DAVINCI_MCASP_RFIFOCTL (0x1018)
+#define DAVINCI_MCASP_RFIFOSTS (0x101C)
+
+/*
+ * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
+ * Register Bits
+ */
+#define MCASP_FREE BIT(0)
+#define MCASP_SOFT BIT(1)
+
+/*
+ * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits
+ */
+#define AXR(n) (1<<n)
+#define PFUNC_AMUTE BIT(25)
+#define ACLKX BIT(26)
+#define AHCLKX BIT(27)
+#define AFSX BIT(28)
+#define ACLKR BIT(29)
+#define AHCLKR BIT(30)
+#define AFSR BIT(31)
+
+/*
+ * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits
+ */
+#define AXR(n) (1<<n)
+#define PDIR_AMUTE BIT(25)
+#define ACLKX BIT(26)
+#define AHCLKX BIT(27)
+#define AFSX BIT(28)
+#define ACLKR BIT(29)
+#define AHCLKR BIT(30)
+#define AFSR BIT(31)
+
+/*
+ * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits
+ */
+#define DITEN BIT(0) /* Transmit DIT mode enable/disable */
+#define VA BIT(2)
+#define VB BIT(3)
+
+/*
+ * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits
+ */
+#define TXROT(val) (val)
+#define TXSEL BIT(3)
+#define TXSSZ(val) (val<<4)
+#define TXPBIT(val) (val<<8)
+#define TXPAD(val) (val<<13)
+#define TXORD BIT(15)
+#define FSXDLY(val) (val<<16)
+
+/*
+ * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits
+ */
+#define RXROT(val) (val)
+#define RXSEL BIT(3)
+#define RXSSZ(val) (val<<4)
+#define RXPBIT(val) (val<<8)
+#define RXPAD(val) (val<<13)
+#define RXORD BIT(15)
+#define FSRDLY(val) (val<<16)
+
+/*
+ * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits
+ */
+#define FSXPOL BIT(0)
+#define AFSXE BIT(1)
+#define FSXDUR BIT(4)
+#define FSXMOD(val) (val<<7)
+
+/*
+ * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits
+ */
+#define FSRPOL BIT(0)
+#define AFSRE BIT(1)
+#define FSRDUR BIT(4)
+#define FSRMOD(val) (val<<7)
+
+/*
+ * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits
+ */
+#define ACLKXDIV(val) (val)
+#define ACLKXE BIT(5)
+#define TX_ASYNC BIT(6)
+#define ACLKXPOL BIT(7)
+
+/*
+ * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
+ */
+#define ACLKRDIV(val) (val)
+#define ACLKRE BIT(5)
+#define RX_ASYNC BIT(6)
+#define ACLKRPOL BIT(7)
+
+/*
+ * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
+ * Register Bits
+ */
+#define AHCLKXDIV(val) (val)
+#define AHCLKXPOL BIT(14)
+#define AHCLKXE BIT(15)
+
+/*
+ * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
+ * Register Bits
+ */
+#define AHCLKRDIV(val) (val)
+#define AHCLKRPOL BIT(14)
+#define AHCLKRE BIT(15)
+
+/*
+ * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits
+ */
+#define MODE(val) (val)
+#define DISMOD (val)(val<<2)
+#define TXSTATE BIT(4)
+#define RXSTATE BIT(5)
+
+/*
+ * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
+ */
+#define LBEN BIT(0)
+#define LBORD BIT(1)
+#define LBGENMODE(val) (val<<2)
+
+/*
+ * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration
+ */
+#define TXTDMS(n) (1<<n)
+
+/*
+ * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration
+ */
+#define RXTDMS(n) (1<<n)
+
+/*
+ * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits
+ */
+#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */
+#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */
+#define RXSERCLR BIT(2) /* Receiver Serializer Clear */
+#define RXSMRST BIT(3) /* Receiver State Machine Reset */
+#define RXFSRST BIT(4) /* Frame Sync Generator Reset */
+#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */
+#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/
+#define TXSERCLR BIT(10) /* Transmit Serializer Clear */
+#define TXSMRST BIT(11) /* Transmitter State Machine Reset */
+#define TXFSRST BIT(12) /* Frame Sync Generator Reset */
+
+/*
+ * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits
+ */
+#define MUTENA(val) (val)
+#define MUTEINPOL BIT(2)
+#define MUTEINENA BIT(3)
+#define MUTEIN BIT(4)
+#define MUTER BIT(5)
+#define MUTEX BIT(6)
+#define MUTEFSR BIT(7)
+#define MUTEFSX BIT(8)
+#define MUTEBADCLKR BIT(9)
+#define MUTEBADCLKX BIT(10)
+#define MUTERXDMAERR BIT(11)
+#define MUTETXDMAERR BIT(12)
+
+/*
+ * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits
+ */
+#define RXDATADMADIS BIT(0)
+
+/*
+ * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits
+ */
+#define TXDATADMADIS BIT(0)
+
+/*
+ * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
+ */
+#define FIFO_ENABLE BIT(16)
+#define NUMEVT_MASK (0xFF << 8)
+#define NUMDMA_MASK (0xFF)
+
+#define DAVINCI_MCASP_NUM_SERIALIZER 16
+
+static inline void mcasp_set_bits(void __iomem *reg, u32 val)
+{
+ __raw_writel(__raw_readl(reg) | val, reg);
+}
+
+static inline void mcasp_clr_bits(void __iomem *reg, u32 val)
+{
+ __raw_writel((__raw_readl(reg) & ~(val)), reg);
+}
+
+static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask)
+{
+ __raw_writel((__raw_readl(reg) & ~mask) | val, reg);
+}
+
+static inline void mcasp_set_reg(void __iomem *reg, u32 val)
+{
+ __raw_writel(val, reg);
+}
+
+static inline u32 mcasp_get_reg(void __iomem *reg)
+{
+ return (unsigned int)__raw_readl(reg);
+}
+
+static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
+{
+ int i = 0;
+
+ mcasp_set_bits(regs, val);
+
+ /* programming GBLCTL needs to read back from GBLCTL and verfiy */
+ /* loop count is to avoid the lock-up */
+ for (i = 0; i < 1000; i++) {
+ if ((mcasp_get_reg(regs) & val) == val)
+ break;
+ }
+
+ if (i == 1000 && ((mcasp_get_reg(regs) & val) != val))
+ printk(KERN_ERR "GBLCTL write error\n");
+}
+
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct davinci_audio_dev *dev = cpu_dai->private_data;
+ cpu_dai->dma_data = dev->dma_params[substream->stream];
+ return 0;
+}
+
+static void mcasp_start_rx(struct davinci_audio_dev *dev)
+{
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
+
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
+
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+}
+
+static void mcasp_start_tx(struct davinci_audio_dev *dev)
+{
+ u8 offset = 0, i;
+ u32 cnt;
+
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
+
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
+ mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
+ for (i = 0; i < dev->num_serializer; i++) {
+ if (dev->serial_dir[i] == TX_MODE) {
+ offset = i;
+ break;
+ }
+ }
+
+ /* wait for TX ready */
+ cnt = 0;
+ while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) &
+ TXSTATE) && (cnt < 100000))
+ cnt++;
+
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
+}
+
+static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
+{
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mcasp_start_tx(dev);
+ else
+ mcasp_start_rx(dev);
+
+ /* enable FIFO */
+ if (dev->txnumevt)
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
+
+ if (dev->rxnumevt)
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+}
+
+static void mcasp_stop_rx(struct davinci_audio_dev *dev)
+{
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+}
+
+static void mcasp_stop_tx(struct davinci_audio_dev *dev)
+{
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+}
+
+static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
+{
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mcasp_stop_tx(dev);
+ else
+ mcasp_stop_rx(dev);
+
+ /* disable FIFO */
+ if (dev->txnumevt)
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
+
+ if (dev->rxnumevt)
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+}
+
+static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct davinci_audio_dev *dev = cpu_dai->private_data;
+ void __iomem *base = dev->base;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /* codec is clock and frame slave */
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+ mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+ mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26));
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ /* codec is clock master and frame slave */
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+ mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+ mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26));
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* codec is clock and frame master */
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+ mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+ mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26));
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+ mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+ mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+ break;
+
+ case SND_SOC_DAIFMT_NB_IF:
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+ mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+ mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+ break;
+
+ case SND_SOC_DAIFMT_IB_IF:
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+ mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+ mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+ break;
+
+ case SND_SOC_DAIFMT_NB_NF:
+ mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+ mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+
+ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+ mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int davinci_config_channel_size(struct davinci_audio_dev *dev,
+ int channel_size)
+{
+ u32 fmt = 0;
+
+ switch (channel_size) {
+ case DAVINCI_AUDIO_WORD_8:
+ fmt = 0x03;
+ break;
+
+ case DAVINCI_AUDIO_WORD_12:
+ fmt = 0x05;
+ break;
+
+ case DAVINCI_AUDIO_WORD_16:
+ fmt = 0x07;
+ break;
+
+ case DAVINCI_AUDIO_WORD_20:
+ fmt = 0x09;
+ break;
+
+ case DAVINCI_AUDIO_WORD_24:
+ fmt = 0x0B;
+ break;
+
+ case DAVINCI_AUDIO_WORD_28:
+ fmt = 0x0D;
+ break;
+
+ case DAVINCI_AUDIO_WORD_32:
+ fmt = 0x0F;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
+ RXSSZ(fmt), RXSSZ(0x0F));
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+ TXSSZ(fmt), TXSSZ(0x0F));
+ return 0;
+}
+
+static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
+{
+ int i;
+ u8 tx_ser = 0;
+ u8 rx_ser = 0;
+
+ /* Default configuration */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
+
+ /* All PINS as McASP */
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG,
+ TXDATADMADIS);
+ } else {
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG,
+ RXDATADMADIS);
+ }
+
+ for (i = 0; i < dev->num_serializer; i++) {
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
+ dev->serial_dir[i]);
+ if (dev->serial_dir[i] == TX_MODE) {
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
+ AXR(i));
+ tx_ser++;
+ } else if (dev->serial_dir[i] == RX_MODE) {
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
+ AXR(i));
+ rx_ser++;
+ }
+ }
+
+ if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt * tx_ser > 64)
+ dev->txnumevt = 1;
+
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser,
+ NUMDMA_MASK);
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
+ }
+
+ if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (dev->rxnumevt * rx_ser > 64)
+ dev->rxnumevt = 1;
+
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser,
+ NUMDMA_MASK);
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
+}
+
+static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
+{
+ int i, active_slots;
+ u32 mask = 0;
+
+ active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots;
+ for (i = 0; i < active_slots; i++)
+ mask |= (1 << i);
+
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* bit stream is MSB first with no delay */
+ /* DSP_B mode */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG,
+ AHCLKXE);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
+
+ if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
+ else
+ printk(KERN_ERR "playback tdm slot %d not supported\n",
+ dev->tdm_slots);
+
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+ } else {
+ /* bit stream is MSB first with no delay */
+ /* DSP_B mode */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG,
+ AHCLKRE);
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
+
+ if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
+ else
+ printk(KERN_ERR "capture tdm slot %d not supported\n",
+ dev->tdm_slots);
+
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+ }
+}
+
+/* S/PDIF */
+static void davinci_hw_dit_param(struct davinci_audio_dev *dev)
+{
+ /* Set the PDIR for Serialiser as output */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX);
+
+ /* TXMASK for 24 bits */
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF);
+
+ /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
+ and LSB first */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+ TXROT(6) | TXSSZ(15));
+
+ /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
+ AFSXE | FSXMOD(0x180));
+
+ /* Set the TX tdm : for all the slots */
+ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
+
+ /* Set the TX clock controls : div = 1 and internal */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
+ ACLKXE | TX_ASYNC);
+
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+
+ /* Only 44100 and 48000 are valid, both have the same setting */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
+
+ /* Enable the DIT */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+}
+
+static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct davinci_audio_dev *dev = cpu_dai->private_data;
+ struct davinci_pcm_dma_params *dma_params =
+ dev->dma_params[substream->stream];
+ int word_length;
+ u8 numevt;
+
+ davinci_hw_common_param(dev, substream->stream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ numevt = dev->txnumevt;
+ else
+ numevt = dev->rxnumevt;
+
+ if (!numevt)
+ numevt = 1;
+
+ if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
+ davinci_hw_dit_param(dev);
+ else
+ davinci_hw_param(dev, substream->stream);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ dma_params->data_type = 1;
+ word_length = DAVINCI_AUDIO_WORD_8;
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ dma_params->data_type = 2;
+ word_length = DAVINCI_AUDIO_WORD_16;
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dma_params->data_type = 4;
+ word_length = DAVINCI_AUDIO_WORD_32;
+ break;
+
+ default:
+ printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
+ return -EINVAL;
+ }
+
+ if (dev->version == MCASP_VERSION_2) {
+ dma_params->data_type *= numevt;
+ dma_params->acnt = 4 * numevt;
+ } else
+ dma_params->acnt = dma_params->data_type;
+
+ davinci_config_channel_size(dev, word_length);
+
+ return 0;
+}
+
+static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ davinci_mcasp_start(dev, substream->stream);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ davinci_mcasp_stop(dev, substream->stream);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+ .startup = davinci_mcasp_startup,
+ .trigger = davinci_mcasp_trigger,
+ .hw_params = davinci_mcasp_hw_params,
+ .set_fmt = davinci_mcasp_set_dai_fmt,
+
+};
+
+struct snd_soc_dai davinci_mcasp_dai[] = {
+ {
+ .name = "davinci-i2s",
+ .id = 0,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_MCASP_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_MCASP_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &davinci_mcasp_dai_ops,
+
+ },
+ {
+ .name = "davinci-dit",
+ .id = 1,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = DAVINCI_MCASP_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &davinci_mcasp_dai_ops,
+ },
+
+};
+EXPORT_SYMBOL_GPL(davinci_mcasp_dai);
+
+static int davinci_mcasp_probe(struct platform_device *pdev)
+{
+ struct davinci_pcm_dma_params *dma_data;
+ struct resource *mem, *ioarea, *res;
+ struct snd_platform_data *pdata;
+ struct davinci_audio_dev *dev;
+ int count = 0;
+ int ret = 0;
+
+ dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
+ if (!dev)
+ return -ENOMEM;
+
+ dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
+ GFP_KERNEL);
+ if (!dma_data) {
+ ret = -ENOMEM;
+ goto err_release_dev;
+ }
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ ret = -ENODEV;
+ goto err_release_data;
+ }
+
+ ioarea = request_mem_region(mem->start,
+ (mem->end - mem->start) + 1, pdev->name);
+ if (!ioarea) {
+ dev_err(&pdev->dev, "Audio region already claimed\n");
+ ret = -EBUSY;
+ goto err_release_data;
+ }
+
+ pdata = pdev->dev.platform_data;
+ dev->clk = clk_get(&pdev->dev, NULL);
+ if (IS_ERR(dev->clk)) {
+ ret = -ENODEV;
+ goto err_release_region;
+ }
+
+ clk_enable(dev->clk);
+
+ dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+ dev->op_mode = pdata->op_mode;
+ dev->tdm_slots = pdata->tdm_slots;
+ dev->num_serializer = pdata->num_serializer;
+ dev->serial_dir = pdata->serial_dir;
+ dev->codec_fmt = pdata->codec_fmt;
+ dev->version = pdata->version;
+ dev->txnumevt = pdata->txnumevt;
+ dev->rxnumevt = pdata->rxnumevt;
+
+ dma_data[count].name = "I2S PCM Stereo out";
+ dma_data[count].eventq_no = pdata->eventq_no;
+ dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+ io_v2p(dev->base));
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
+
+ /* first TX, then RX */
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!res) {
+ dev_err(&pdev->dev, "no DMA resource\n");
+ goto err_release_region;
+ }
+
+ dma_data[count].channel = res->start;
+ count++;
+ dma_data[count].name = "I2S PCM Stereo in";
+ dma_data[count].eventq_no = pdata->eventq_no;
+ dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+ io_v2p(dev->base));
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
+
+ res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!res) {
+ dev_err(&pdev->dev, "no DMA resource\n");
+ goto err_release_region;
+ }
+
+ dma_data[count].channel = res->start;
+ davinci_mcasp_dai[pdata->op_mode].private_data = dev;
+ davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
+ ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
+
+ if (ret != 0)
+ goto err_release_region;
+ return 0;
+
+err_release_region:
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+err_release_data:
+ kfree(dma_data);
+err_release_dev:
+ kfree(dev);
+
+ return ret;
+}
+
+static int davinci_mcasp_remove(struct platform_device *pdev)
+{
+ struct snd_platform_data *pdata = pdev->dev.platform_data;
+ struct davinci_pcm_dma_params *dma_data;
+ struct davinci_audio_dev *dev;
+ struct resource *mem;
+
+ snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]);
+ dev = davinci_mcasp_dai[pdata->op_mode].private_data;
+ clk_disable(dev->clk);
+ clk_put(dev->clk);
+ dev->clk = NULL;
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+
+ dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ kfree(dma_data);
+ kfree(dev);
+
+ return 0;
+}
+
+static struct platform_driver davinci_mcasp_driver = {
+ .probe = davinci_mcasp_probe,
+ .remove = davinci_mcasp_remove,
+ .driver = {
+ .name = "davinci-mcasp",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init davinci_mcasp_init(void)
+{
+ return platform_driver_register(&davinci_mcasp_driver);
+}
+module_init(davinci_mcasp_init);
+
+static void __exit davinci_mcasp_exit(void)
+{
+ platform_driver_unregister(&davinci_mcasp_driver);
+}
+module_exit(davinci_mcasp_exit);
+
+MODULE_AUTHOR("Steve Chen");
+MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
new file mode 100644
index 0000000..554354c
--- /dev/null
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -0,0 +1,60 @@
+/*
+ * ALSA SoC McASP Audio Layer for TI DAVINCI processor
+ *
+ * MCASP related definitions
+ *
+ * Author: Nirmal Pandey <n-pandey@ti.com>,
+ * Suresh Rajashekara <suresh.r@ti.com>
+ * Steve Chen <schen@.mvista.com>
+ *
+ * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright: (C) 2009 Texas Instruments, India
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef DAVINCI_MCASP_H
+#define DAVINCI_MCASP_H
+
+#include <linux/io.h>
+#include <mach/asp.h>
+#include "davinci-pcm.h"
+
+extern struct snd_soc_dai davinci_mcasp_dai[];
+
+#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000
+#define DAVINCI_MCASP_I2S_DAI 0
+#define DAVINCI_MCASP_DIT_DAI 1
+
+enum {
+ DAVINCI_AUDIO_WORD_8 = 0,
+ DAVINCI_AUDIO_WORD_12,
+ DAVINCI_AUDIO_WORD_16,
+ DAVINCI_AUDIO_WORD_20,
+ DAVINCI_AUDIO_WORD_24,
+ DAVINCI_AUDIO_WORD_32,
+ DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */
+};
+
+struct davinci_audio_dev {
+ void __iomem *base;
+ int sample_rate;
+ struct clk *clk;
+ struct davinci_pcm_dma_params *dma_params[2];
+ unsigned int codec_fmt;
+
+ /* McASP specific data */
+ int tdm_slots;
+ u8 op_mode;
+ u8 num_serializer;
+ u8 *serial_dir;
+ u8 version;
+
+ /* McASP FIFO related */
+ u8 txnumevt;
+ u8 rxnumevt;
+};
+
+#endif /* DAVINCI_MCASP_H */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index a059965..091dacb 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
dma_addr_t src, dst;
unsigned short src_bidx, dst_bidx;
unsigned int data_type;
+ unsigned short acnt;
unsigned int count;
period_size = snd_pcm_lib_period_bytes(substream);
@@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
dst_bidx = data_type;
}
+ acnt = prtd->params->acnt;
edma_set_src(lch, src, INCR, W8BIT);
edma_set_dest(lch, dst, INCR, W8BIT);
edma_set_src_index(lch, src_bidx, 0);
edma_set_dest_index(lch, dst_bidx, 0);
- edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+ edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
@@ -206,6 +208,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
/* Copy self-linked parameter RAM entry into master channel */
edma_read_slot(prtd->slave_lch, &temp);
edma_write_slot(prtd->master_lch, &temp);
+ davinci_pcm_enqueue_dma(substream);
return 0;
}
@@ -243,6 +246,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
int ret = 0;
snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ return ret;
prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL);
if (prtd == NULL)
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 62cb4eb..63d9625 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -12,17 +12,20 @@
#ifndef _DAVINCI_PCM_H
#define _DAVINCI_PCM_H
+#include <mach/edma.h>
+#include <mach/asp.h>
+
+
struct davinci_pcm_dma_params {
- char *name; /* stream identifier */
- int channel; /* sync dma channel ID */
- dma_addr_t dma_addr; /* device physical address for DMA */
- unsigned int data_type; /* xfer data type */
+ char *name; /* stream identifier */
+ int channel; /* sync dma channel ID */
+ unsigned short acnt;
+ dma_addr_t dma_addr; /* device physical address for DMA */
+ enum dma_event_q eventq_no; /* event queue number */
+ unsigned char data_type; /* xfer data type */
+ unsigned char convert_mono_stereo;
};
-struct evm_snd_platform_data {
- int tx_dma_ch;
- int rx_dma_ch;
-};
extern struct snd_soc_platform davinci_soc_platform;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index f0a2d40..9ff62e3 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -69,6 +69,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
{
+ if (s->appl_ptr > s->runtime->control->appl_ptr) {
+ /*
+ * In this case s->runtime->control->appl_ptr has wrapped around.
+ * Play the data to the end of the boundary, then wrap our own
+ * appl_ptr back around.
+ */
+ while (s->appl_ptr < s->runtime->boundary) {
+ if (bcom_queue_full(s->bcom_task))
+ return;
+
+ s->appl_ptr += s->period_size;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+ s->appl_ptr -= s->runtime->boundary;
+ }
+
while (s->appl_ptr < s->runtime->control->appl_ptr) {
if (bcom_queue_full(s->bcom_task))
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 7eb5499..c4ae3e0 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -12,6 +12,7 @@
#include <linux/module.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
+#include <linux/delay.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -112,7 +113,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
out_8(&regs->op1, MPC52xx_PSC_OP_RES);
udelay(10);
out_8(&regs->op0, MPC52xx_PSC_OP_RES);
- udelay(50);
+ msleep(1);
psc_ac97_warm_reset(ac97);
}
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
new file mode 100644
index 0000000..a700562e
--- /dev/null
+++ b/sound/soc/imx/Kconfig
@@ -0,0 +1,21 @@
+config SND_MX1_MX2_SOC
+ tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs"
+ depends on ARCH_MX2 || ARCH_MX1
+ select SND_PCM
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the MX1 or MX2 SSI interface.
+
+config SND_MXC_SOC_SSI
+ tristate
+
+config SND_SOC_MX27VIS_WM8974
+ tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board"
+ depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10
+ select SND_MXC_SOC_SSI
+ select SND_SOC_WM8974
+ help
+ Say Y if you want to add support for SoC audio on Visstrim SM10
+ board with WM8974.
+
+
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
new file mode 100644
index 0000000..c2ffd2c
--- /dev/null
+++ b/sound/soc/imx/Makefile
@@ -0,0 +1,10 @@
+# i.MX Platform Support
+snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o
+snd-soc-mxc-ssi-objs := mxc-ssi.o
+
+obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o
+obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o
+
+# i.MX Machine Support
+snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o
+obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o
diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c
new file mode 100644
index 0000000..b838665
--- /dev/null
+++ b/sound/soc/imx/mx1_mx2-pcm.c
@@ -0,0 +1,488 @@
+/*
+ * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs
+ *
+ * Copyright 2009 Vista Silicon S.L.
+ * Author: Javier Martin
+ * javier.martin@vista-silicon.com
+ *
+ * Based on mxc-pcm.c by Liam Girdwood.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/dma.h>
+#include <mach/hardware.h>
+#include <mach/dma-mx1-mx2.h>
+
+#include "mx1_mx2-pcm.h"
+
+
+static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .buffer_bytes_max = 32 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+struct mx1_mx2_runtime_data {
+ int dma_ch;
+ int active;
+ unsigned int period;
+ unsigned int periods;
+ int tx_spin;
+ spinlock_t dma_lock;
+ struct mx1_mx2_pcm_dma_params *dma_params;
+};
+
+
+/**
+ * This function stops the current dma transfer for playback
+ * and clears the dma pointers.
+ *
+ * @param substream pointer to the structure of the current stream.
+ *
+ */
+static int audio_stop_dma(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&prtd->dma_lock, flags);
+
+ pr_debug("%s\n", __func__);
+
+ prtd->active = 0;
+ prtd->period = 0;
+ prtd->periods = 0;
+
+ /* this stops the dma channel and clears the buffer ptrs */
+
+ imx_dma_disable(prtd->dma_ch);
+
+ spin_unlock_irqrestore(&prtd->dma_lock, flags);
+
+ return 0;
+}
+
+/**
+ * This function is called whenever a new audio block needs to be
+ * transferred to the codec. The function receives the address and the size
+ * of the new block and start a new DMA transfer.
+ *
+ * @param substream pointer to the structure of the current stream.
+ *
+ */
+static int dma_new_period(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+ unsigned int dma_size;
+ unsigned int offset;
+ int ret = 0;
+ dma_addr_t mem_addr;
+ unsigned int dev_addr;
+
+ if (prtd->active) {
+ dma_size = frames_to_bytes(runtime, runtime->period_size);
+ offset = dma_size * prtd->period;
+
+ pr_debug("%s: period (%d) out of (%d)\n", __func__,
+ prtd->period,
+ runtime->periods);
+ pr_debug("period_size %d frames\n offset %d bytes\n",
+ (unsigned int)runtime->period_size,
+ offset);
+ pr_debug("dma_size %d bytes\n", dma_size);
+
+ snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max);
+
+ mem_addr = (dma_addr_t)(runtime->dma_addr + offset);
+ dev_addr = prtd->dma_params->per_address;
+ pr_debug("%s: mem_addr is %x\n dev_addr is %x\n",
+ __func__, mem_addr, dev_addr);
+
+ ret = imx_dma_setup_single(prtd->dma_ch, mem_addr,
+ dma_size, dev_addr,
+ prtd->dma_params->transfer_type);
+ if (ret < 0) {
+ printk(KERN_ERR "Error %d configuring DMA\n", ret);
+ return ret;
+ }
+ imx_dma_enable(prtd->dma_ch);
+
+ pr_debug("%s: transfer enabled\nmem_addr = %x\n",
+ __func__, (unsigned int) mem_addr);
+ pr_debug("dev_addr = %x\ndma_size = %d\n",
+ (unsigned int) dev_addr, dma_size);
+
+ prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */
+ prtd->period++;
+ prtd->period %= runtime->periods;
+ }
+ return ret;
+}
+
+
+/**
+ * This is a callback which will be called
+ * when a TX transfer finishes. The call occurs
+ * in interrupt context.
+ *
+ * @param dat pointer to the structure of the current stream.
+ *
+ */
+static void audio_dma_irq(int channel, void *data)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ struct mx1_mx2_runtime_data *prtd;
+ unsigned int dma_size;
+ unsigned int previous_period;
+ unsigned int offset;
+
+ substream = data;
+ runtime = substream->runtime;
+ prtd = runtime->private_data;
+ previous_period = prtd->periods;
+ dma_size = frames_to_bytes(runtime, runtime->period_size);
+ offset = dma_size * previous_period;
+
+ prtd->tx_spin = 0;
+ prtd->periods++;
+ prtd->periods %= runtime->periods;
+
+ pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset);
+
+ /*
+ * If we are getting a callback for an active stream then we inform
+ * the PCM middle layer we've finished a period
+ */
+ if (prtd->active)
+ snd_pcm_period_elapsed(substream);
+
+ /*
+ * Trig next DMA transfer
+ */
+ dma_new_period(substream);
+}
+
+/**
+ * This function configures the hardware to allow audio
+ * playback operations. It is called by ALSA framework.
+ *
+ * @param substream pointer to the structure of the current stream.
+ *
+ * @return 0 on success, -1 otherwise.
+ */
+static int
+snd_mx1_mx2_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+
+ prtd->period = 0;
+ prtd->periods = 0;
+
+ return 0;
+}
+
+static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0) {
+ printk(KERN_ERR "%s: Error %d failed to malloc pcm pages \n",
+ __func__, ret);
+ return ret;
+ }
+
+ pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_addr);
+ pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_area);
+ pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_bytes);
+
+ return ret;
+}
+
+static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+
+ imx_dma_free(prtd->dma_ch);
+
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->tx_spin = 0;
+ /* requested stream startup */
+ prtd->active = 1;
+ pr_debug("%s: starting dma_new_period\n", __func__);
+ ret = dma_new_period(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ /* requested stream shutdown */
+ pr_debug("%s: stopping dma transfer\n", __func__);
+ ret = audio_stop_dma(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+ unsigned int offset = 0;
+
+ /* tx_spin value is used here to check if a transfer is active */
+ if (prtd->tx_spin) {
+ offset = (runtime->period_size * (prtd->periods)) +
+ (runtime->period_size >> 1);
+ if (offset >= runtime->buffer_size)
+ offset = runtime->period_size >> 1;
+ } else {
+ offset = (runtime->period_size * (prtd->periods));
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+ }
+ pr_debug("%s: pointer offset %x\n", __func__, offset);
+
+ return offset;
+}
+
+static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ return ret;
+
+ prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ runtime->private_data = prtd;
+
+ if (!dma_data)
+ return -ENODEV;
+
+ prtd->dma_params = dma_data;
+
+ pr_debug("%s: Requesting dma channel (%s)\n", __func__,
+ prtd->dma_params->name);
+ prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name,
+ DMA_PRIO_HIGH);
+ if (prtd->dma_ch < 0) {
+ printk(KERN_ERR "Error %d requesting dma channel\n", ret);
+ return ret;
+ }
+ imx_dma_config_burstlen(prtd->dma_ch,
+ prtd->dma_params->watermark_level);
+
+ ret = imx_dma_config_channel(prtd->dma_ch,
+ prtd->dma_params->per_config,
+ prtd->dma_params->mem_config,
+ prtd->dma_params->event_id, 0);
+
+ if (ret) {
+ pr_debug(KERN_ERR "Error %d configuring dma channel %d\n",
+ ret, prtd->dma_ch);
+ return ret;
+ }
+
+ pr_debug("%s: Setting tx dma callback function\n", __func__);
+ ret = imx_dma_setup_handlers(prtd->dma_ch,
+ audio_dma_irq, NULL,
+ (void *)substream);
+ if (ret < 0) {
+ printk(KERN_ERR "Error %d setting dma callback function\n", ret);
+ return ret;
+ }
+ return 0;
+
+ out:
+ return ret;
+}
+
+static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct mx1_mx2_runtime_data *prtd = runtime->private_data;
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops mx1_mx2_pcm_ops = {
+ .open = mx1_mx2_pcm_open,
+ .close = mx1_mx2_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = mx1_mx2_pcm_hw_params,
+ .hw_free = mx1_mx2_pcm_hw_free,
+ .prepare = snd_mx1_mx2_prepare,
+ .trigger = mx1_mx2_pcm_trigger,
+ .pointer = mx1_mx2_pcm_pointer,
+ .mmap = mx1_mx2_pcm_mmap,
+};
+
+static u64 mx1_mx2_pcm_dmamask = 0xffffffff;
+
+static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max;
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+
+ /* Reserve uncached-buffered memory area for DMA */
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ __func__, (void *) buf->area, (void *) buf->addr, size);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &mx1_mx2_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ pr_debug("%s: preallocate playback buffer\n", __func__);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ pr_debug("%s: preallocate capture buffer\n", __func__);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform mx1_mx2_soc_platform = {
+ .name = "mx1_mx2-audio",
+ .pcm_ops = &mx1_mx2_pcm_ops,
+ .pcm_new = mx1_mx2_pcm_new,
+ .pcm_free = mx1_mx2_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform);
+
+static int __init mx1_mx2_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&mx1_mx2_soc_platform);
+}
+module_init(mx1_mx2_soc_platform_init);
+
+static void __exit mx1_mx2_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&mx1_mx2_soc_platform);
+}
+module_exit(mx1_mx2_soc_platform_exit);
+
+MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com");
+MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h
new file mode 100644
index 0000000..2e52810
--- /dev/null
+++ b/sound/soc/imx/mx1_mx2-pcm.h
@@ -0,0 +1,26 @@
+/*
+ * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _MX1_MX2_PCM_H
+#define _MX1_MX2_PCM_H
+
+/* DMA information for mx1_mx2 platforms */
+struct mx1_mx2_pcm_dma_params {
+ char *name; /* stream identifier */
+ unsigned int transfer_type; /* READ or WRITE DMA transfer */
+ dma_addr_t per_address; /* physical address of SSI fifo */
+ int event_id; /* fixed DMA number for SSI fifo */
+ int watermark_level; /* SSI fifo watermark level */
+ int per_config; /* DMA Config flags for peripheral */
+ int mem_config; /* DMA Config flags for RAM */
+ };
+
+/* platform data */
+extern struct snd_soc_platform mx1_mx2_soc_platform;
+
+#endif
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
new file mode 100644
index 0000000..e4dcb53
--- /dev/null
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -0,0 +1,317 @@
+/*
+ * mx27vis_wm8974.c -- SoC audio for mx27vis
+ *
+ * Copyright 2009 Vista Silicon S.L.
+ * Author: Javier Martin
+ * javier.martin@vista-silicon.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+
+#include "../codecs/wm8974.h"
+#include "mx1_mx2-pcm.h"
+#include "mxc-ssi.h"
+#include <mach/gpio.h>
+#include <mach/iomux.h>
+
+#define IGNORED_ARG 0
+
+
+static struct snd_soc_card mx27vis;
+
+/**
+ * This function connects SSI1 (HPCR1) as slave to
+ * SSI1 external signals (PPCR1)
+ * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from
+ * port 4
+ */
+void audmux_connect_1_4(void)
+{
+ pr_debug("AUDMUX: normal operation mode\n");
+ /* Reset HPCR1 and PPCR1 */
+
+ DAM_HPCR1 = 0x00000000;
+ DAM_PPCR1 = 0x00000000;
+
+ /* set to synchronous */
+ DAM_HPCR1 |= AUDMUX_HPCR_SYN;
+ DAM_PPCR1 |= AUDMUX_PPCR_SYN;
+
+
+ /* set Rx sources 1 <--> 4 */
+ DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */
+ DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */
+
+ /* set Tx frame and Clock direction and source 4 --> 1 output */
+ DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR;
+ DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */
+
+ return;
+}
+
+static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0;
+ int ret = 0;
+
+ /*
+ * The WM8974 is better at generating accurate audio clocks than the
+ * MX27 SSI controller, so we will use it as master when we can.
+ */
+ switch (params_rate(params)) {
+ case 8000:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ mclk = WM8974_MCLKDIV_12;
+ pll_out = 24576000;
+ break;
+ case 16000:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ pll_out = 12288000;
+ break;
+ case 48000:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_4;
+ pll_out = 12288000;
+ break;
+ case 96000:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_2;
+ pll_out = 12288000;
+ break;
+ case 11025:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_16;
+ pll_out = 11289600;
+ break;
+ case 22050:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_8;
+ pll_out = 11289600;
+ break;
+ case 44100:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_4;
+ mclk = WM8974_MCLKDIV_2;
+ pll_out = 11289600;
+ break;
+ case 88200:
+ fmt = SND_SOC_DAIFMT_CBM_CFM;
+ bclk = WM8974_BCLKDIV_2;
+ pll_out = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = codec_dai->ops->set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_SYNC | fmt);
+ if (ret < 0) {
+ printk(KERN_ERR "Error from codec DAI configuration\n");
+ return ret;
+ }
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->ops->set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_SYNC | fmt);
+ if (ret < 0) {
+ printk(KERN_ERR "Error from cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Put DC field of STCCR to 1 (not zero) */
+ ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2);
+
+ /* set the SSI system clock as input */
+ ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "Error when setting system SSI clk\n");
+ return ret;
+ }
+
+ /* set codec BCLK division for sample rate */
+ ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk);
+ if (ret < 0) {
+ printk(KERN_ERR "Error when setting BCLK division\n");
+ return ret;
+ }
+
+
+ /* codec PLL input is 25 MHz */
+ ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG,
+ 25000000, pll_out);
+ if (ret < 0) {
+ printk(KERN_ERR "Error when setting PLL input\n");
+ return ret;
+ }
+
+ /*set codec MCLK division for sample rate */
+ ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk);
+ if (ret < 0) {
+ printk(KERN_ERR "Error when setting MCLK division\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0);
+}
+
+/*
+ * mx27vis WM8974 HiFi DAI opserations.
+ */
+static struct snd_soc_ops mx27vis_hifi_ops = {
+ .hw_params = mx27vis_hifi_hw_params,
+ .hw_free = mx27vis_hifi_hw_free,
+};
+
+
+static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ return 0;
+}
+
+static int mx27vis_resume(struct platform_device *pdev)
+{
+ return 0;
+}
+
+static int mx27vis_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+
+ ret = get_ssi_clk(0, &pdev->dev);
+
+ if (ret < 0) {
+ printk(KERN_ERR "%s: cant get ssi clock\n", __func__);
+ return ret;
+ }
+
+
+ return 0;
+}
+
+static int mx27vis_remove(struct platform_device *pdev)
+{
+ put_ssi_clk(0);
+ return 0;
+}
+
+static struct snd_soc_dai_link mx27vis_dai[] = {
+{ /* Hifi Playback*/
+ .name = "WM8974",
+ .stream_name = "WM8974 HiFi",
+ .cpu_dai = &imx_ssi_pcm_dai[0],
+ .codec_dai = &wm8974_dai,
+ .ops = &mx27vis_hifi_ops,
+},
+};
+
+static struct snd_soc_card mx27vis = {
+ .name = "mx27vis",
+ .platform = &mx1_mx2_soc_platform,
+ .probe = mx27vis_probe,
+ .remove = mx27vis_remove,
+ .suspend_pre = mx27vis_suspend,
+ .resume_post = mx27vis_resume,
+ .dai_link = mx27vis_dai,
+ .num_links = ARRAY_SIZE(mx27vis_dai),
+};
+
+static struct snd_soc_device mx27vis_snd_devdata = {
+ .card = &mx27vis,
+ .codec_dev = &soc_codec_dev_wm8974,
+};
+
+static struct platform_device *mx27vis_snd_device;
+
+/* Temporal definition of board specific behaviour */
+void gpio_ssi_active(int ssi_num)
+{
+ int ret = 0;
+
+ unsigned int ssi1_pins[] = {
+ PC20_PF_SSI1_FS,
+ PC21_PF_SSI1_RXD,
+ PC22_PF_SSI1_TXD,
+ PC23_PF_SSI1_CLK,
+ };
+ unsigned int ssi2_pins[] = {
+ PC24_PF_SSI2_FS,
+ PC25_PF_SSI2_RXD,
+ PC26_PF_SSI2_TXD,
+ PC27_PF_SSI2_CLK,
+ };
+ if (ssi_num == 0)
+ ret = mxc_gpio_setup_multiple_pins(ssi1_pins,
+ ARRAY_SIZE(ssi1_pins), "USB OTG");
+ else
+ ret = mxc_gpio_setup_multiple_pins(ssi2_pins,
+ ARRAY_SIZE(ssi2_pins), "USB OTG");
+ if (ret)
+ printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num);
+}
+
+
+static int __init mx27vis_init(void)
+{
+ int ret;
+
+ mx27vis_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!mx27vis_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata);
+ mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev;
+ ret = platform_device_add(mx27vis_snd_device);
+
+ if (ret) {
+ printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+ platform_device_put(mx27vis_snd_device);
+ }
+
+ /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */
+ gpio_ssi_active(0);
+ audmux_connect_1_4();
+
+ return ret;
+}
+
+static void __exit mx27vis_exit(void)
+{
+ /* We should call some "ssi_gpio_inactive()" properly */
+}
+
+module_init(mx27vis_init);
+module_exit(mx27vis_exit);
+
+
+MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com");
+MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
new file mode 100644
index 0000000..3806ff2
--- /dev/null
+++ b/sound/soc/imx/mxc-ssi.c
@@ -0,0 +1,868 @@
+/*
+ * mxc-ssi.c -- SSI driver for Freescale IMX
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * Based on mxc-alsa-mc13783 (C) 2006 Freescale.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * TODO:
+ * Need to rework SSI register defs when new defs go into mainline.
+ * Add support for TDM and FIFO 1.
+ * Add support for i.mx3x DMA interface.
+ *
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/dma-mx1-mx2.h>
+#include <asm/mach-types.h>
+
+#include "mxc-ssi.h"
+#include "mx1_mx2-pcm.h"
+
+#define SSI1_PORT 0
+#define SSI2_PORT 1
+
+static int ssi_active[2] = {0, 0};
+
+/* DMA information for mx1_mx2 platforms */
+static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = {
+ .name = "SSI1 PCM Stereo out 0",
+ .transfer_type = DMA_MODE_WRITE,
+ .per_address = SSI1_BASE_ADDR + STX0,
+ .event_id = DMA_REQ_SSI1_TX0,
+ .watermark_level = TXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = {
+ .name = "SSI1 PCM Stereo out 1",
+ .transfer_type = DMA_MODE_WRITE,
+ .per_address = SSI1_BASE_ADDR + STX1,
+ .event_id = DMA_REQ_SSI1_TX1,
+ .watermark_level = TXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = {
+ .name = "SSI1 PCM Stereo in 0",
+ .transfer_type = DMA_MODE_READ,
+ .per_address = SSI1_BASE_ADDR + SRX0,
+ .event_id = DMA_REQ_SSI1_RX0,
+ .watermark_level = RXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = {
+ .name = "SSI1 PCM Stereo in 1",
+ .transfer_type = DMA_MODE_READ,
+ .per_address = SSI1_BASE_ADDR + SRX1,
+ .event_id = DMA_REQ_SSI1_RX1,
+ .watermark_level = RXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = {
+ .name = "SSI2 PCM Stereo out 0",
+ .transfer_type = DMA_MODE_WRITE,
+ .per_address = SSI2_BASE_ADDR + STX0,
+ .event_id = DMA_REQ_SSI2_TX0,
+ .watermark_level = TXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = {
+ .name = "SSI2 PCM Stereo out 1",
+ .transfer_type = DMA_MODE_WRITE,
+ .per_address = SSI2_BASE_ADDR + STX1,
+ .event_id = DMA_REQ_SSI2_TX1,
+ .watermark_level = TXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = {
+ .name = "SSI2 PCM Stereo in 0",
+ .transfer_type = DMA_MODE_READ,
+ .per_address = SSI2_BASE_ADDR + SRX0,
+ .event_id = DMA_REQ_SSI2_RX0,
+ .watermark_level = RXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = {
+ .name = "SSI2 PCM Stereo in 1",
+ .transfer_type = DMA_MODE_READ,
+ .per_address = SSI2_BASE_ADDR + SRX1,
+ .event_id = DMA_REQ_SSI2_RX1,
+ .watermark_level = RXFIFO_WATERMARK,
+ .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
+ .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
+};
+
+static struct clk *ssi_clk0, *ssi_clk1;
+
+int get_ssi_clk(int ssi, struct device *dev)
+{
+ switch (ssi) {
+ case 0:
+ ssi_clk0 = clk_get(dev, "ssi1");
+ if (IS_ERR(ssi_clk0))
+ return PTR_ERR(ssi_clk0);
+ return 0;
+ case 1:
+ ssi_clk1 = clk_get(dev, "ssi2");
+ if (IS_ERR(ssi_clk1))
+ return PTR_ERR(ssi_clk1);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+EXPORT_SYMBOL(get_ssi_clk);
+
+void put_ssi_clk(int ssi)
+{
+ switch (ssi) {
+ case 0:
+ clk_put(ssi_clk0);
+ ssi_clk0 = NULL;
+ break;
+ case 1:
+ clk_put(ssi_clk1);
+ ssi_clk1 = NULL;
+ break;
+ }
+}
+EXPORT_SYMBOL(put_ssi_clk);
+
+/*
+ * SSI system clock configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ u32 scr;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ scr = SSI1_SCR;
+ pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr);
+ } else {
+ scr = SSI2_SCR;
+ pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr);
+ }
+
+ if (scr & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+
+ switch (clk_id) {
+ case IMX_SSP_SYS_CLK:
+ if (dir == SND_SOC_CLOCK_OUT) {
+ scr |= SSI_SCR_SYS_CLK_EN;
+ pr_debug("%s: clk of is output\n", __func__);
+ } else {
+ scr &= ~SSI_SCR_SYS_CLK_EN;
+ pr_debug("%s: clk of is input\n", __func__);
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ pr_debug("%s: writeback of SSI1_SCR\n", __func__);
+ SSI1_SCR = scr;
+ } else {
+ pr_debug("%s: writeback of SSI2_SCR\n", __func__);
+ SSI2_SCR = scr;
+ }
+
+ return 0;
+}
+
+/*
+ * SSI Clock dividers
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ u32 stccr, srccr;
+
+ pr_debug("%s\n", __func__);
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ if (SSI1_SCR & SSI_SCR_SSIEN)
+ return 0;
+ srccr = SSI1_STCCR;
+ stccr = SSI1_STCCR;
+ } else {
+ if (SSI2_SCR & SSI_SCR_SSIEN)
+ return 0;
+ srccr = SSI2_STCCR;
+ stccr = SSI2_STCCR;
+ }
+
+ switch (div_id) {
+ case IMX_SSI_TX_DIV_2:
+ stccr &= ~SSI_STCCR_DIV2;
+ stccr |= div;
+ break;
+ case IMX_SSI_TX_DIV_PSR:
+ stccr &= ~SSI_STCCR_PSR;
+ stccr |= div;
+ break;
+ case IMX_SSI_TX_DIV_PM:
+ stccr &= ~0xff;
+ stccr |= SSI_STCCR_PM(div);
+ break;
+ case IMX_SSI_RX_DIV_2:
+ stccr &= ~SSI_STCCR_DIV2;
+ stccr |= div;
+ break;
+ case IMX_SSI_RX_DIV_PSR:
+ stccr &= ~SSI_STCCR_PSR;
+ stccr |= div;
+ break;
+ case IMX_SSI_RX_DIV_PM:
+ stccr &= ~0xff;
+ stccr |= SSI_STCCR_PM(div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ SSI1_STCCR = stccr;
+ SSI1_SRCCR = srccr;
+ } else {
+ SSI2_STCCR = stccr;
+ SSI2_SRCCR = srccr;
+ }
+ return 0;
+}
+
+/*
+ * SSI Network Mode or TDM slots configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ */
+static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int mask, int slots)
+{
+ u32 stmsk, srmsk, stccr;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ if (SSI1_SCR & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+ stccr = SSI1_STCCR;
+ } else {
+ if (SSI2_SCR & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+ stccr = SSI2_STCCR;
+ }
+
+ stmsk = srmsk = mask;
+ stccr &= ~SSI_STCCR_DC_MASK;
+ stccr |= SSI_STCCR_DC(slots - 1);
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ SSI1_STMSK = stmsk;
+ SSI1_SRMSK = srmsk;
+ SSI1_SRCCR = SSI1_STCCR = stccr;
+ } else {
+ SSI2_STMSK = stmsk;
+ SSI2_SRMSK = srmsk;
+ SSI2_SRCCR = SSI2_STCCR = stccr;
+ }
+
+ return 0;
+}
+
+/*
+ * SSI DAI format configuration.
+ * Should only be called when port is inactive (i.e. SSIEN = 0).
+ * Note: We don't use the I2S modes but instead manually configure the
+ * SSI for I2S.
+ */
+static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ u32 stcr = 0, srcr = 0, scr;
+
+ /*
+ * This is done to avoid this function to modify
+ * previous set values in stcr
+ */
+ stcr = SSI1_STCR;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2)
+ scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET);
+ else
+ scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET);
+
+ if (scr & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+
+ /* DAI mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* data on rising edge of bclk, frame low 1clk before data */
+ stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+ srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /* data on rising edge of bclk, frame high with data */
+ stcr |= SSI_STCR_TXBIT0;
+ srcr |= SSI_SRCR_RXBIT0;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /* data on rising edge of bclk, frame high with data */
+ stcr |= SSI_STCR_TFSL;
+ srcr |= SSI_SRCR_RFSL;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* data on rising edge of bclk, frame high 1clk before data */
+ stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS;
+ srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS;
+ break;
+ }
+
+ /* DAI clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ stcr |= SSI_STCR_TFSI;
+ stcr &= ~SSI_STCR_TSCKP;
+ srcr |= SSI_SRCR_RFSI;
+ srcr &= ~SSI_SRCR_RSCKP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+ srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+ srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP;
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ stcr &= ~SSI_STCR_TFSI;
+ stcr |= SSI_STCR_TSCKP;
+ srcr &= ~SSI_SRCR_RFSI;
+ srcr |= SSI_SRCR_RSCKP;
+ break;
+ }
+
+ /* DAI clock master masks */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR;
+ srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ stcr |= SSI_STCR_TFDIR;
+ srcr |= SSI_SRCR_RFDIR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ stcr |= SSI_STCR_TXDIR;
+ srcr |= SSI_SRCR_RXDIR;
+ break;
+ }
+
+ /* sync */
+ if (!(fmt & SND_SOC_DAIFMT_ASYNC))
+ scr |= SSI_SCR_SYN;
+
+ /* tdm - only for stereo atm */
+ if (fmt & SND_SOC_DAIFMT_TDM)
+ scr |= SSI_SCR_NET;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ SSI1_STCR = stcr;
+ SSI1_SRCR = srcr;
+ SSI1_SCR = scr;
+ } else {
+ SSI2_STCR = stcr;
+ SSI2_SRCR = srcr;
+ SSI2_SCR = scr;
+ }
+
+ return 0;
+}
+
+static int imx_ssi_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* set up TX DMA params */
+ switch (cpu_dai->id) {
+ case IMX_DAI_SSI0:
+ cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0;
+ break;
+ case IMX_DAI_SSI1:
+ cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1;
+ break;
+ case IMX_DAI_SSI2:
+ cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0;
+ break;
+ case IMX_DAI_SSI3:
+ cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1;
+ }
+ pr_debug("%s: (playback)\n", __func__);
+ } else {
+ /* set up RX DMA params */
+ switch (cpu_dai->id) {
+ case IMX_DAI_SSI0:
+ cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0;
+ break;
+ case IMX_DAI_SSI1:
+ cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1;
+ break;
+ case IMX_DAI_SSI2:
+ cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0;
+ break;
+ case IMX_DAI_SSI3:
+ cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1;
+ }
+ pr_debug("%s: (capture)\n", __func__);
+ }
+
+ /*
+ * we cant really change any SSI values after SSI is enabled
+ * need to fix in software for max flexibility - lrg
+ */
+ if (cpu_dai->active) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+
+ /* reset the SSI port - Sect 45.4.4 */
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+
+ if (!ssi_clk0)
+ return -EINVAL;
+
+ if (ssi_active[SSI1_PORT]++) {
+ pr_debug("%s: exit before reset\n", __func__);
+ return 0;
+ }
+
+ /* SSI1 Reset */
+ SSI1_SCR = 0;
+
+ SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) |
+ SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) |
+ SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) |
+ SSI_SFCSR_TFWM0(TXFIFO_WATERMARK);
+ } else {
+
+ if (!ssi_clk1)
+ return -EINVAL;
+
+ if (ssi_active[SSI2_PORT]++) {
+ pr_debug("%s: exit before reset\n", __func__);
+ return 0;
+ }
+
+ /* SSI2 Reset */
+ SSI2_SCR = 0;
+
+ SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) |
+ SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) |
+ SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) |
+ SSI_SFCSR_TFWM0(TXFIFO_WATERMARK);
+ }
+
+ return 0;
+}
+
+int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ u32 stccr, stcr, sier;
+
+ pr_debug("%s\n", __func__);
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK;
+ stcr = SSI1_STCR;
+ sier = SSI1_SIER;
+ } else {
+ stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK;
+ stcr = SSI2_STCR;
+ sier = SSI2_SIER;
+ }
+
+ /* DAI data (word) size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ stccr |= SSI_STCCR_WL(16);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ stccr |= SSI_STCCR_WL(20);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ stccr |= SSI_STCCR_WL(24);
+ break;
+ }
+
+ /* enable interrupts */
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2)
+ stcr |= SSI_STCR_TFEN0;
+ else
+ stcr |= SSI_STCR_TFEN1;
+ sier |= SSI_SIER_TDMAE;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ SSI1_STCR = stcr;
+ SSI1_STCCR = stccr;
+ SSI1_SIER = sier;
+ } else {
+ SSI2_STCR = stcr;
+ SSI2_STCCR = stccr;
+ SSI2_SIER = sier;
+ }
+
+ return 0;
+}
+
+int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ u32 srccr, srcr, sier;
+
+ pr_debug("%s\n", __func__);
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK;
+ srcr = SSI1_SRCR;
+ sier = SSI1_SIER;
+ } else {
+ srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK;
+ srcr = SSI2_SRCR;
+ sier = SSI2_SIER;
+ }
+
+ /* DAI data (word) size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ srccr |= SSI_SRCCR_WL(16);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ srccr |= SSI_SRCCR_WL(20);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ srccr |= SSI_SRCCR_WL(24);
+ break;
+ }
+
+ /* enable interrupts */
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2)
+ srcr |= SSI_SRCR_RFEN0;
+ else
+ srcr |= SSI_SRCR_RFEN1;
+ sier |= SSI_SIER_RDMAE;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ SSI1_SRCR = srcr;
+ SSI1_SRCCR = srccr;
+ SSI1_SIER = sier;
+ } else {
+ SSI2_SRCR = srcr;
+ SSI2_SRCCR = srccr;
+ SSI2_SIER = sier;
+ }
+
+ return 0;
+}
+
+/*
+ * Should only be called when port is inactive (i.e. SSIEN = 0),
+ * although can be called multiple times by upper layers.
+ */
+int imx_ssi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ int ret;
+
+ /* cant change any parameters when SSI is running */
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ if (SSI1_SCR & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+ } else {
+ if (SSI2_SCR & SSI_SCR_SSIEN) {
+ printk(KERN_WARNING "Warning ssi already enabled\n");
+ return 0;
+ }
+ }
+
+ /*
+ * Configure both tx and rx params with the same settings. This is
+ * really a harware restriction because SSI must be disabled until
+ * we can change those values. If there is an active audio stream in
+ * one direction, enabling the other direction with different
+ * settings would mean disturbing the running one.
+ */
+ ret = imx_ssi_hw_tx_params(substream, params);
+ if (ret < 0)
+ return ret;
+ return imx_ssi_hw_rx_params(substream, params);
+}
+
+int imx_ssi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+
+ /* Enable clks here to follow SSI recommended init sequence */
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
+ ret = clk_enable(ssi_clk0);
+ if (ret < 0)
+ printk(KERN_ERR "Unable to enable ssi_clk0\n");
+ } else {
+ ret = clk_enable(ssi_clk1);
+ if (ret < 0)
+ printk(KERN_ERR "Unable to enable ssi_clk1\n");
+ }
+
+ return 0;
+}
+
+static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ u32 scr;
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2)
+ scr = SSI1_SCR;
+ else
+ scr = SSI2_SCR;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ scr |= SSI_SCR_TE | SSI_SCR_SSIEN;
+ else
+ scr |= SSI_SCR_RE | SSI_SCR_SSIEN;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ scr &= ~SSI_SCR_TE;
+ else
+ scr &= ~SSI_SCR_RE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2)
+ SSI1_SCR = scr;
+ else
+ SSI2_SCR = scr;
+
+ return 0;
+}
+
+static void imx_ssi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ /* shutdown SSI if neither Tx or Rx is active */
+ if (!cpu_dai->active) {
+
+ if (cpu_dai->id == IMX_DAI_SSI0 ||
+ cpu_dai->id == IMX_DAI_SSI2) {
+
+ if (--ssi_active[SSI1_PORT] > 1)
+ return;
+
+ SSI1_SCR = 0;
+ clk_disable(ssi_clk0);
+ } else {
+ if (--ssi_active[SSI2_PORT])
+ return;
+ SSI2_SCR = 0;
+ clk_disable(ssi_clk1);
+ }
+ }
+}
+
+#ifdef CONFIG_PM
+static int imx_ssi_suspend(struct platform_device *dev,
+ struct snd_soc_dai *dai)
+{
+ return 0;
+}
+
+static int imx_ssi_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ return 0;
+}
+
+#else
+#define imx_ssi_suspend NULL
+#define imx_ssi_resume NULL
+#endif
+
+#define IMX_SSI_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000)
+
+#define IMX_SSI_BITS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = {
+ .startup = imx_ssi_startup,
+ .shutdown = imx_ssi_shutdown,
+ .trigger = imx_ssi_trigger,
+ .prepare = imx_ssi_prepare,
+ .hw_params = imx_ssi_hw_params,
+ .set_sysclk = imx_ssi_set_dai_sysclk,
+ .set_clkdiv = imx_ssi_set_dai_clkdiv,
+ .set_fmt = imx_ssi_set_dai_fmt,
+ .set_tdm_slot = imx_ssi_set_dai_tdm_slot,
+};
+
+struct snd_soc_dai imx_ssi_pcm_dai[] = {
+{
+ .name = "imx-i2s-1-0",
+ .id = IMX_DAI_SSI0,
+ .suspend = imx_ssi_suspend,
+ .resume = imx_ssi_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .ops = &imx_ssi_pcm_dai_ops,
+},
+{
+ .name = "imx-i2s-2-0",
+ .id = IMX_DAI_SSI1,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .ops = &imx_ssi_pcm_dai_ops,
+},
+{
+ .name = "imx-i2s-1-1",
+ .id = IMX_DAI_SSI2,
+ .suspend = imx_ssi_suspend,
+ .resume = imx_ssi_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .ops = &imx_ssi_pcm_dai_ops,
+},
+{
+ .name = "imx-i2s-2-1",
+ .id = IMX_DAI_SSI3,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .formats = IMX_SSI_BITS,
+ .rates = IMX_SSI_RATES,},
+ .ops = &imx_ssi_pcm_dai_ops,
+},
+};
+EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai);
+
+static int __init imx_ssi_init(void)
+{
+ return snd_soc_register_dais(imx_ssi_pcm_dai,
+ ARRAY_SIZE(imx_ssi_pcm_dai));
+}
+
+static void __exit imx_ssi_exit(void)
+{
+ snd_soc_unregister_dais(imx_ssi_pcm_dai,
+ ARRAY_SIZE(imx_ssi_pcm_dai));
+}
+
+module_init(imx_ssi_init);
+module_exit(imx_ssi_exit);
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com");
+MODULE_DESCRIPTION("i.MX ASoC I2S driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h
new file mode 100644
index 0000000..12bbdc9
--- /dev/null
+++ b/sound/soc/imx/mxc-ssi.h
@@ -0,0 +1,238 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _IMX_SSI_H
+#define _IMX_SSI_H
+
+#include <mach/hardware.h>
+
+/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */
+#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR)
+#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR)
+
+#define STX0 0x00
+#define STX1 0x04
+#define SRX0 0x08
+#define SRX1 0x0c
+#define SCR 0x10
+#define SISR 0x14
+#define SIER 0x18
+#define STCR 0x1c
+#define SRCR 0x20
+#define STCCR 0x24
+#define SRCCR 0x28
+#define SFCSR 0x2c
+#define STR 0x30
+#define SOR 0x34
+#define SACNT 0x38
+#define SACADD 0x3c
+#define SACDAT 0x40
+#define SATAG 0x44
+#define STMSK 0x48
+#define SRMSK 0x4c
+
+#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0)))
+#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1)))
+#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0)))
+#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1)))
+#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR)))
+#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR)))
+#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER)))
+#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR)))
+#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR)))
+#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR)))
+#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR)))
+#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR)))
+#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR)))
+#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR)))
+#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT)))
+#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD)))
+#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT)))
+#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG)))
+#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK)))
+#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK)))
+
+
+#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0)))
+#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1)))
+#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0)))
+#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1)))
+#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR)))
+#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR)))
+#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER)))
+#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR)))
+#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR)))
+#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR)))
+#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR)))
+#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR)))
+#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR)))
+#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR)))
+#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT)))
+#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD)))
+#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT)))
+#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG)))
+#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK)))
+#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK)))
+
+#define SSI_SCR_CLK_IST (1 << 9)
+#define SSI_SCR_TCH_EN (1 << 8)
+#define SSI_SCR_SYS_CLK_EN (1 << 7)
+#define SSI_SCR_I2S_MODE_NORM (0 << 5)
+#define SSI_SCR_I2S_MODE_MSTR (1 << 5)
+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
+#define SSI_SCR_SYN (1 << 4)
+#define SSI_SCR_NET (1 << 3)
+#define SSI_SCR_RE (1 << 2)
+#define SSI_SCR_TE (1 << 1)
+#define SSI_SCR_SSIEN (1 << 0)
+
+#define SSI_SISR_CMDAU (1 << 18)
+#define SSI_SISR_CMDDU (1 << 17)
+#define SSI_SISR_RXT (1 << 16)
+#define SSI_SISR_RDR1 (1 << 15)
+#define SSI_SISR_RDR0 (1 << 14)
+#define SSI_SISR_TDE1 (1 << 13)
+#define SSI_SISR_TDE0 (1 << 12)
+#define SSI_SISR_ROE1 (1 << 11)
+#define SSI_SISR_ROE0 (1 << 10)
+#define SSI_SISR_TUE1 (1 << 9)
+#define SSI_SISR_TUE0 (1 << 8)
+#define SSI_SISR_TFS (1 << 7)
+#define SSI_SISR_RFS (1 << 6)
+#define SSI_SISR_TLS (1 << 5)
+#define SSI_SISR_RLS (1 << 4)
+#define SSI_SISR_RFF1 (1 << 3)
+#define SSI_SISR_RFF0 (1 << 2)
+#define SSI_SISR_TFE1 (1 << 1)
+#define SSI_SISR_TFE0 (1 << 0)
+
+#define SSI_SIER_RDMAE (1 << 22)
+#define SSI_SIER_RIE (1 << 21)
+#define SSI_SIER_TDMAE (1 << 20)
+#define SSI_SIER_TIE (1 << 19)
+#define SSI_SIER_CMDAU_EN (1 << 18)
+#define SSI_SIER_CMDDU_EN (1 << 17)
+#define SSI_SIER_RXT_EN (1 << 16)
+#define SSI_SIER_RDR1_EN (1 << 15)
+#define SSI_SIER_RDR0_EN (1 << 14)
+#define SSI_SIER_TDE1_EN (1 << 13)
+#define SSI_SIER_TDE0_EN (1 << 12)
+#define SSI_SIER_ROE1_EN (1 << 11)
+#define SSI_SIER_ROE0_EN (1 << 10)
+#define SSI_SIER_TUE1_EN (1 << 9)
+#define SSI_SIER_TUE0_EN (1 << 8)
+#define SSI_SIER_TFS_EN (1 << 7)
+#define SSI_SIER_RFS_EN (1 << 6)
+#define SSI_SIER_TLS_EN (1 << 5)
+#define SSI_SIER_RLS_EN (1 << 4)
+#define SSI_SIER_RFF1_EN (1 << 3)
+#define SSI_SIER_RFF0_EN (1 << 2)
+#define SSI_SIER_TFE1_EN (1 << 1)
+#define SSI_SIER_TFE0_EN (1 << 0)
+
+#define SSI_STCR_TXBIT0 (1 << 9)
+#define SSI_STCR_TFEN1 (1 << 8)
+#define SSI_STCR_TFEN0 (1 << 7)
+#define SSI_STCR_TFDIR (1 << 6)
+#define SSI_STCR_TXDIR (1 << 5)
+#define SSI_STCR_TSHFD (1 << 4)
+#define SSI_STCR_TSCKP (1 << 3)
+#define SSI_STCR_TFSI (1 << 2)
+#define SSI_STCR_TFSL (1 << 1)
+#define SSI_STCR_TEFS (1 << 0)
+
+#define SSI_SRCR_RXBIT0 (1 << 9)
+#define SSI_SRCR_RFEN1 (1 << 8)
+#define SSI_SRCR_RFEN0 (1 << 7)
+#define SSI_SRCR_RFDIR (1 << 6)
+#define SSI_SRCR_RXDIR (1 << 5)
+#define SSI_SRCR_RSHFD (1 << 4)
+#define SSI_SRCR_RSCKP (1 << 3)
+#define SSI_SRCR_RFSI (1 << 2)
+#define SSI_SRCR_RFSL (1 << 1)
+#define SSI_SRCR_REFS (1 << 0)
+
+#define SSI_STCCR_DIV2 (1 << 18)
+#define SSI_STCCR_PSR (1 << 15)
+#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13)
+#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8)
+#define SSI_STCCR_PM(x) (((x) & 0xff) << 0)
+#define SSI_STCCR_WL_MASK (0xf << 13)
+#define SSI_STCCR_DC_MASK (0x1f << 8)
+#define SSI_STCCR_PM_MASK (0xff << 0)
+
+#define SSI_SRCCR_DIV2 (1 << 18)
+#define SSI_SRCCR_PSR (1 << 15)
+#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13)
+#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8)
+#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0)
+#define SSI_SRCCR_WL_MASK (0xf << 13)
+#define SSI_SRCCR_DC_MASK (0x1f << 8)
+#define SSI_SRCCR_PM_MASK (0xff << 0)
+
+
+#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28)
+#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24)
+#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20)
+#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16)
+#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12)
+#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8)
+#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4)
+#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0)
+
+#define SSI_STR_TEST (1 << 15)
+#define SSI_STR_RCK2TCK (1 << 14)
+#define SSI_STR_RFS2TFS (1 << 13)
+#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8)
+#define SSI_STR_TXD2RXD (1 << 7)
+#define SSI_STR_TCK2RCK (1 << 6)
+#define SSI_STR_TFS2RFS (1 << 5)
+#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0)
+
+#define SSI_SOR_CLKOFF (1 << 6)
+#define SSI_SOR_RX_CLR (1 << 5)
+#define SSI_SOR_TX_CLR (1 << 4)
+#define SSI_SOR_INIT (1 << 3)
+#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1)
+#define SSI_SOR_SYNRST (1 << 0)
+
+#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5)
+#define SSI_SACNT_WR (x << 4)
+#define SSI_SACNT_RD (x << 3)
+#define SSI_SACNT_TIF (x << 2)
+#define SSI_SACNT_FV (x << 1)
+#define SSI_SACNT_AC97EN (x << 0)
+
+/* Watermarks for FIFO's */
+#define TXFIFO_WATERMARK 0x4
+#define RXFIFO_WATERMARK 0x4
+
+/* i.MX DAI SSP ID's */
+#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */
+#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */
+#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */
+#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */
+
+/* SSI clock sources */
+#define IMX_SSP_SYS_CLK 0
+
+/* SSI audio dividers */
+#define IMX_SSI_TX_DIV_2 0
+#define IMX_SSI_TX_DIV_PSR 1
+#define IMX_SSI_TX_DIV_PM 2
+#define IMX_SSI_RX_DIV_2 3
+#define IMX_SSI_RX_DIV_PSR 4
+#define IMX_SSI_RX_DIV_PM 5
+
+
+/* SSI Div 2 */
+#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2)
+#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2
+
+extern struct snd_soc_dai imx_ssi_pcm_dai[4];
+extern int get_ssi_clk(int ssi, struct device *dev);
+extern void put_ssi_clk(int ssi);
+#endif
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b771238..2dee983 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810
help
Say Y if you want to add support for SoC audio on Nokia N810.
+config SND_OMAP_SOC_AMS_DELTA
+ tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+ depends on SND_OMAP_SOC && MACH_AMS_DELTA
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_CX20442
+ help
+ Say Y if you want to add support for SoC audio on Amstrad Delta.
+
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
@@ -72,4 +80,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE
help
Say Y if you want to add support for SoC audio on the Beagleboard.
+config SND_OMAP_SOC_ZOOM2
+ tristate "SoC Audio support for Zoom2"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for Soc audio on Zoom2 board.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index a37f498..02d6947 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
+snd-soc-ams-delta-objs := ams-delta.o
snd-soc-osk5912-objs := osk5912.o
snd-soc-overo-objs := overo.o
snd-soc-omap2evm-objs := omap2evm.o
@@ -14,8 +15,10 @@ snd-soc-omap3evm-objs := omap3evm.o
snd-soc-sdp3430-objs := sdp3430.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
+snd-soc-zoom2-objs := zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
@@ -23,3 +26,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
+obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 0000000..5a5166a
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,646 @@
+/*
+ * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
+ *
+ * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/omap/osk5912.x
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/gpio.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/cx20442.h"
+
+
+/* Board specific DAPM widgets */
+ const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+ /* Handset */
+ SND_SOC_DAPM_MIC("Mouthpiece", NULL),
+ SND_SOC_DAPM_HP("Earpiece", NULL),
+ /* Handsfree/Speakerphone */
+ SND_SOC_DAPM_MIC("Microphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* How they are connected to codec pins */
+static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
+ {"TELIN", NULL, "Mouthpiece"},
+ {"Earpiece", NULL, "TELOUT"},
+
+ {"MIC", NULL, "Microphone"},
+ {"Speaker", NULL, "SPKOUT"},
+};
+
+/*
+ * Controls, functional after the modem line discipline is activated.
+ */
+
+/* Virtual switch: audio input/output constellations */
+static const char *ams_delta_audio_mode[] =
+ {"Mixed", "Handset", "Handsfree", "Speakerphone"};
+
+/* Selection <-> pin translation */
+#define AMS_DELTA_MOUTHPIECE 0
+#define AMS_DELTA_EARPIECE 1
+#define AMS_DELTA_MICROPHONE 2
+#define AMS_DELTA_SPEAKER 3
+#define AMS_DELTA_AGC 4
+
+#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
+ (1 << AMS_DELTA_MICROPHONE))
+#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
+ (1 << AMS_DELTA_EARPIECE))
+#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
+ (1 << AMS_DELTA_SPEAKER))
+#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
+
+unsigned short ams_delta_audio_mode_pins[] = {
+ AMS_DELTA_MIXED,
+ AMS_DELTA_HANDSET,
+ AMS_DELTA_HANDSFREE,
+ AMS_DELTA_SPEAKERPHONE,
+};
+
+static unsigned short ams_delta_audio_agc;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
+ unsigned short pins;
+ int pin, changed = 0;
+
+ /* Refuse any mode changes if we are not able to control the codec. */
+ if (!codec->control_data)
+ return -EUNATCH;
+
+ if (ucontrol->value.enumerated.item[0] >= control->max)
+ return -EINVAL;
+
+ mutex_lock(&codec->mutex);
+
+ /* Translate selection to bitmap */
+ pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
+
+ /* Setup pins after corresponding bits if changed */
+ pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+ else
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ else
+ snd_soc_dapm_disable_pin(codec, "Earpiece");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ else
+ snd_soc_dapm_disable_pin(codec, "Microphone");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_AGC));
+ if (pin != ams_delta_audio_agc) {
+ ams_delta_audio_agc = pin;
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "AGCIN");
+ else
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ }
+ if (changed)
+ snd_soc_dapm_sync(codec);
+
+ mutex_unlock(&codec->mutex);
+
+ return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned short pins, mode;
+
+ pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+ AMS_DELTA_MOUTHPIECE) |
+ (snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+ AMS_DELTA_EARPIECE));
+ if (pins)
+ pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ AMS_DELTA_MICROPHONE);
+ else
+ pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ AMS_DELTA_MICROPHONE) |
+ (snd_soc_dapm_get_pin_status(codec, "Speaker") <<
+ AMS_DELTA_SPEAKER) |
+ (ams_delta_audio_agc << AMS_DELTA_AGC));
+
+ for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
+ if (pins == ams_delta_audio_mode_pins[mode])
+ break;
+
+ if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
+ return -EINVAL;
+
+ ucontrol->value.enumerated.item[0] = mode;
+
+ return 0;
+}
+
+static const struct soc_enum ams_delta_audio_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
+ ams_delta_audio_mode),
+};
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+ SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+ ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
+
+/* Hook switch */
+static struct snd_soc_jack ams_delta_hook_switch;
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
+ {
+ .gpio = 4,
+ .name = "hook_switch",
+ .report = SND_JACK_HEADSET,
+ .invert = 1,
+ .debounce_time = 150,
+ }
+};
+
+/* After we are able to control the codec over the modem,
+ * the hook switch can be used for dynamic DAPM reconfiguration. */
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+ /* Handset */
+ {
+ .pin = "Mouthpiece",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Earpiece",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ /* Handsfree */
+ {
+ .pin = "Microphone",
+ .mask = SND_JACK_MICROPHONE,
+ .invert = 1,
+ },
+ {
+ .pin = "Speaker",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+};
+
+
+/*
+ * Modem line discipline, required for making above controls functional.
+ * Activated from userspace with ldattach, possibly invoked from udev rule.
+ */
+
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment. Be carefull not
+ * to interfere with our digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending;
+static bool ams_delta_muted;
+static DEFINE_SPINLOCK(ams_delta_lock);
+
+static void cx81801_timeout(unsigned long data)
+{
+ int muted;
+
+ spin_lock(&ams_delta_lock);
+ cx81801_cmd_pending = 0;
+ muted = ams_delta_muted;
+ spin_unlock(&ams_delta_lock);
+
+ /* Reconnect the codec DAI back from the modem to the CPU DAI
+ * only if digital mute still off */
+ if (!muted)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
+}
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+ return v253_ops.open(tty);
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+
+ del_timer_sync(&cx81801_timer);
+
+ v253_ops.close(tty);
+
+ /* Prevent the hook switch from further changing the DAPM pins */
+ INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+ /* Revert back to default audio input/output constellation */
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ snd_soc_dapm_sync(codec);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+ cx81801_close(tty);
+ return 0;
+}
+
+/* Line discipline .recieve_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+ const unsigned char *cp, char *fp, int count)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+ const unsigned char *c;
+ int apply, ret;
+
+ if (!codec->control_data) {
+ /* First modem response, complete setup procedure */
+
+ /* Initialize timer used for config pulse generation */
+ setup_timer(&cx81801_timer, cx81801_timeout, 0);
+
+ v253_ops.receive_buf(tty, cp, fp, count);
+
+ /* Link hook switch to DAPM pins */
+ ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_pins),
+ ams_delta_hook_switch_pins);
+ if (ret)
+ dev_warn(codec->socdev->card->dev,
+ "Failed to link hook switch to DAPM pins, "
+ "will continue with hook switch unlinked.\n");
+
+ return;
+ }
+
+ v253_ops.receive_buf(tty, cp, fp, count);
+
+ for (c = &cp[count - 1]; c >= cp; c--) {
+ if (*c != '\r')
+ continue;
+ /* Complete modem response received, apply config to codec */
+
+ spin_lock_bh(&ams_delta_lock);
+ mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
+ apply = !ams_delta_muted && !cx81801_cmd_pending;
+ cx81801_cmd_pending = 1;
+ spin_unlock_bh(&ams_delta_lock);
+
+ /* Apply config pulse by connecting the codec to the modem
+ * if not already done */
+ if (apply)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+ AMS_DELTA_LATCH2_MODEM_CODEC);
+ break;
+ }
+}
+
+/* Line discipline .write_wakeup() */
+static void cx81801_wakeup(struct tty_struct *tty)
+{
+ v253_ops.write_wakeup(tty);
+}
+
+static struct tty_ldisc_ops cx81801_ops = {
+ .magic = TTY_LDISC_MAGIC,
+ .name = "cx81801",
+ .owner = THIS_MODULE,
+ .open = cx81801_open,
+ .close = cx81801_close,
+ .hangup = cx81801_hangup,
+ .receive_buf = cx81801_receive,
+ .write_wakeup = cx81801_wakeup,
+};
+
+
+/*
+ * Even if not very usefull, the sound card can still work without any of the
+ * above functonality activated. You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issueing AT commands
+ * over the modem port.
+ */
+
+static int ams_delta_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* Set cpu DAI configuration */
+ return snd_soc_dai_set_fmt(rtd->dai->cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+}
+
+static struct snd_soc_ops ams_delta_ops = {
+ .hw_params = ams_delta_hw_params,
+};
+
+
+/* Board specific codec bias level control */
+static int ams_delta_set_bias_level(struct snd_soc_card *card,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_codec *codec = card->codec;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+ AMS_DELTA_LATCH2_MODEM_NRESET);
+ break;
+ case SND_SOC_BIAS_OFF:
+ if (codec->bias_level != SND_SOC_BIAS_OFF)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+ 0);
+ }
+ codec->bias_level = level;
+
+ return 0;
+}
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ int apply;
+
+ if (ams_delta_muted == mute)
+ return 0;
+
+ spin_lock_bh(&ams_delta_lock);
+ ams_delta_muted = mute;
+ apply = !cx81801_cmd_pending;
+ spin_unlock_bh(&ams_delta_lock);
+
+ if (apply)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+ mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
+ return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static struct snd_soc_dai_ops ams_delta_dai_ops = {
+ .digital_mute = ams_delta_digital_mute,
+};
+
+/* Will be used if the codec ever has its own digital_mute function */
+static int ams_delta_startup(struct snd_pcm_substream *substream)
+{
+ return ams_delta_digital_mute(NULL, 0);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+ ams_delta_digital_mute(NULL, 1);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dai *codec_dai = codec->dai;
+ struct snd_soc_card *card = codec->socdev->card;
+ int ret;
+ /* Codec is ready, now add/activate board specific controls */
+
+ /* Set up digital mute if not provided by the codec */
+ if (!codec_dai->ops) {
+ codec_dai->ops = &ams_delta_dai_ops;
+ } else if (!codec_dai->ops->digital_mute) {
+ codec_dai->ops->digital_mute = ams_delta_digital_mute;
+ } else {
+ ams_delta_ops.startup = ams_delta_startup;
+ ams_delta_ops.shutdown = ams_delta_shutdown;
+ }
+
+ /* Set codec bias level */
+ ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+
+ /* Add hook switch - can be used to control the codec from userspace
+ * even if line discipline fails */
+ ret = snd_soc_jack_new(card, "hook_switch",
+ SND_JACK_HEADSET, &ams_delta_hook_switch);
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to allocate resources for hook switch, "
+ "will continue without one.\n");
+ else {
+ ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to set up hook switch GPIO line, "
+ "will continue with hook switch inactive.\n");
+ }
+
+ /* Register optional line discipline for over the modem control */
+ ret = tty_register_ldisc(N_V253, &cx81801_ops);
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to register line discipline, "
+ "will continue without any controls.\n");
+ return 0;
+ }
+
+ /* Add board specific DAPM widgets and routes */
+ ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+ ARRAY_SIZE(ams_delta_dapm_widgets));
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to register DAPM controls, "
+ "will continue without any.\n");
+ return 0;
+ }
+
+ ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+ ARRAY_SIZE(ams_delta_audio_map));
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to set up DAPM routes, "
+ "will continue with codec default map.\n");
+ return 0;
+ }
+
+ /* Set up initial pin constellation */
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ snd_soc_dapm_disable_pin(codec, "AGCOUT");
+ snd_soc_dapm_sync(codec);
+
+ /* Add virtual switch */
+ ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+ ARRAY_SIZE(ams_delta_audio_controls));
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to register audio mode control, "
+ "will continue without it.\n");
+
+ return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+ .name = "CX20442",
+ .stream_name = "CX20442",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &cx20442_dai,
+ .init = ams_delta_cx20442_init,
+ .ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+ .name = "AMS_DELTA",
+ .platform = &omap_soc_platform,
+ .dai_link = &ams_delta_dai_link,
+ .num_links = 1,
+ .set_bias_level = ams_delta_set_bias_level,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device ams_delta_snd_soc_device = {
+ .card = &ams_delta_audio_card,
+ .codec_dev = &cx20442_codec_dev,
+};
+
+/* Module init/exit */
+static struct platform_device *ams_delta_audio_platform_device;
+static struct platform_device *cx20442_platform_device;
+
+static int __init ams_delta_module_init(void)
+{
+ int ret;
+
+ if (!(machine_is_ams_delta()))
+ return -ENODEV;
+
+ ams_delta_audio_platform_device =
+ platform_device_alloc("soc-audio", -1);
+ if (!ams_delta_audio_platform_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(ams_delta_audio_platform_device,
+ &ams_delta_snd_soc_device);
+ ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev;
+ *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1;
+
+ ret = platform_device_add(ams_delta_audio_platform_device);
+ if (ret)
+ goto err;
+
+ /*
+ * Codec platform device could be registered from elsewhere (board?),
+ * but I do it here as it makes sense only if used with the card.
+ */
+ cx20442_platform_device = platform_device_register_simple("cx20442",
+ -1, NULL, 0);
+ return 0;
+err:
+ platform_device_put(ams_delta_audio_platform_device);
+ return ret;
+}
+module_init(ams_delta_module_init);
+
+static void __exit ams_delta_module_exit(void)
+{
+ struct snd_soc_codec *codec;
+ struct tty_struct *tty;
+
+ if (ams_delta_audio_card.codec) {
+ codec = ams_delta_audio_card.codec;
+
+ if (codec->control_data) {
+ tty = codec->control_data;
+
+ tty_hangup(tty);
+ }
+ }
+
+ if (tty_unregister_ldisc(N_V253) != 0)
+ dev_warn(&ams_delta_audio_platform_device->dev,
+ "failed to unregister V253 line discipline\n");
+
+ snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
+
+ /* Keep modem power on */
+ ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+
+ platform_device_unregister(cx20442_platform_device);
+ platform_device_unregister(ams_delta_audio_platform_device);
+}
+module_exit(ams_delta_module_exit);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index b60b1df..0a50593 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -22,6 +22,7 @@
*/
#include <linux/clk.h>
+#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -322,8 +323,6 @@ static struct snd_soc_card snd_soc_n810 = {
/* Audio private data */
static struct aic3x_setup_data n810_aic33_setup = {
- .i2c_bus = 2,
- .i2c_address = 0x18,
.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
};
@@ -337,6 +336,13 @@ static struct snd_soc_device n810_snd_devdata = {
static struct platform_device *n810_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic3x", 0x1b), }
+};
+
static int __init n810_soc_init(void)
{
int err;
@@ -345,6 +351,8 @@ static int __init n810_soc_init(void)
if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
return -ENODEV;
+ i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
+
n810_snd_device = platform_device_alloc("soc-audio", -1);
if (!n810_snd_device)
return -ENOMEM;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index a5d46a7..3341f49 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -139,27 +139,67 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
+ int samples;
+
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ samples = snd_pcm_lib_period_bytes(substream) >> 1;
+ else
+ samples = 1;
+
+ /* Configure McBSP internal buffer usage */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+ else
+ omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+}
+
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int bus_id = mcbsp_data->bus_id;
int err = 0;
- if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ if (!cpu_dai->active)
+ err = omap_mcbsp_request(bus_id);
+
+ if (cpu_is_omap343x()) {
+ int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
+ int max_period;
+
/*
* McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
* Set constraint for minimum buffer size to the same than FIFO
* size in order to avoid underruns in playback startup because
* HW is keeping the DMA request active until FIFO is filled.
*/
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
- }
+ if (bus_id == 1)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 4096, UINT_MAX);
- if (!cpu_dai->active)
- err = omap_mcbsp_request(mcbsp_data->bus_id);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
+ else
+ max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
+
+ max_period++;
+ max_period <<= 1;
+
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ 32, max_period);
+ }
return err;
}
@@ -183,21 +223,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
- int err = 0;
+ int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- if (!mcbsp_data->active++)
- omap_mcbsp_start(mcbsp_data->bus_id);
+ mcbsp_data->active++;
+ omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- if (!--mcbsp_data->active)
- omap_mcbsp_stop(mcbsp_data->bus_id);
+ omap_mcbsp_stop(mcbsp_data->bus_id, play, !play);
+ mcbsp_data->active--;
break;
default:
err = -EINVAL;
@@ -215,7 +255,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
- int wlen, channels, wpf;
+ int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
unsigned long port;
unsigned int format;
@@ -231,6 +271,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
+ omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
+ omap_mcbsp_set_threshold;
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+ MCBSP_DMA_MODE_THRESHOLD)
+ sync_mode = OMAP_DMA_SYNC_FRAME;
} else {
return -ENODEV;
}
@@ -238,6 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+ omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
if (mcbsp_data->configured) {
@@ -321,11 +368,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* Generic McBSP register settings */
regs->spcr2 |= XINTM(3) | FREE;
regs->spcr1 |= RINTM(3);
- regs->rcr2 |= RFIG;
- regs->xcr2 |= XFIG;
+ /* RFIG and XFIG are not defined in 34xx */
+ if (!cpu_is_omap34xx()) {
+ regs->rcr2 |= RFIG;
+ regs->xcr2 |= XFIG;
+ }
if (cpu_is_omap2430() || cpu_is_omap34xx()) {
- regs->xccr = DXENDLY(1) | XDMAEN;
- regs->rccr = RFULL_CYCLE | RDMAEN;
+ regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+ regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -462,6 +512,40 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
return 0;
}
+static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data,
+ int clk_id)
+{
+ int sel_bit, set = 0;
+ u16 reg = OMAP2_CONTROL_DEVCONF0;
+
+ if (cpu_class_is_omap1())
+ return -EINVAL; /* TODO: Can this be implemented for OMAP1? */
+ if (mcbsp_data->bus_id != 0)
+ return -EINVAL;
+
+ switch (clk_id) {
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ set = 1;
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ sel_bit = 3;
+ break;
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ set = 1;
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ sel_bit = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (set)
+ omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
+ else
+ omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+
+ return 0;
+}
+
static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq,
int dir)
@@ -484,6 +568,13 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
break;
+
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id);
+ break;
default:
err = -ENODEV;
}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index c8147aa..647d2f9 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -32,6 +32,10 @@ enum omap_mcbsp_clksrg_clk {
OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */
+ OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */
+ OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */
};
/* McBSP dividers */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 84a1950..5735945 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -59,16 +59,31 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
struct omap_runtime_data *prtd = runtime->private_data;
unsigned long flags;
- if (cpu_is_omap1510()) {
+ if ((cpu_is_omap1510()) &&
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
/*
- * OMAP1510 doesn't support DMA chaining so have to restart
- * the transfer after all periods are transferred
+ * OMAP1510 doesn't fully support DMA progress counter
+ * and there is no software emulation implemented yet,
+ * so have to maintain our own playback progress counter
+ * that can be used by omap_pcm_pointer() instead.
*/
spin_lock_irqsave(&prtd->lock, flags);
+ if ((stat == OMAP_DMA_LAST_IRQ) &&
+ (prtd->period_index == runtime->periods - 1)) {
+ /* we are in sync, do nothing */
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return;
+ }
if (prtd->period_index >= 0) {
- if (++prtd->period_index == runtime->periods) {
+ if (stat & OMAP_DMA_BLOCK_IRQ) {
+ /* end of buffer reached, loop back */
+ prtd->period_index = 0;
+ } else if (stat & OMAP_DMA_LAST_IRQ) {
+ /* update the counter for the last period */
+ prtd->period_index = runtime->periods - 1;
+ } else if (++prtd->period_index >= runtime->periods) {
+ /* end of buffer missed? loop back */
prtd->period_index = 0;
- omap_start_dma(prtd->dma_ch);
}
}
spin_unlock_irqrestore(&prtd->lock, flags);
@@ -100,7 +115,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
prtd->dma_data = dma_data;
err = omap_request_dma(dma_data->dma_req, dma_data->name,
omap_pcm_dma_irq, substream, &prtd->dma_ch);
- if (!err && !cpu_is_omap1510()) {
+ if (!err) {
/*
* Link channel with itself so DMA doesn't need any
* reprogramming while looping the buffer
@@ -119,8 +134,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
if (prtd->dma_data == NULL)
return 0;
- if (!cpu_is_omap1510())
- omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+ omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
omap_free_dma(prtd->dma_ch);
prtd->dma_data = NULL;
@@ -148,7 +162,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
*/
dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
dma_params.trigger = dma_data->dma_req;
- dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ dma_params.sync_mode = dma_data->sync_mode;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
@@ -174,7 +188,15 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
dma_params.frame_count = runtime->periods;
omap_set_dma_params(prtd->dma_ch, &dma_params);
- omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+ if ((cpu_is_omap1510()) &&
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
+ OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
+ else
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+ omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
return 0;
}
@@ -183,6 +205,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = prtd->dma_data;
unsigned long flags;
int ret = 0;
@@ -192,6 +215,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->period_index = 0;
+ /* Configure McBSP internal buffer usage */
+ if (dma_data->set_threshold)
+ dma_data->set_threshold(substream);
+
omap_start_dma(prtd->dma_ch);
break;
@@ -288,7 +315,7 @@ static struct snd_pcm_ops omap_pcm_ops = {
.mmap = omap_pcm_mmap,
};
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
int stream)
@@ -330,7 +357,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
}
}
-int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
@@ -338,7 +365,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->dma_mask)
card->dev->dma_mask = &omap_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
if (dai->playback.channels_min) {
ret = omap_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 8d9d269..38a821d 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,6 +29,8 @@ struct omap_pcm_dma_data {
char *name; /* stream identifier */
int dma_req; /* DMA request line */
unsigned long port_addr; /* transmit/receive register */
+ int sync_mode; /* DMA sync mode */
+ void (*set_threshold)(struct snd_pcm_substream *substream);
};
extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index b719e5d..4a3f62d 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,6 +24,7 @@
#include <linux/clk.h>
#include <linux/platform_device.h>
+#include <linux/i2c/twl4030.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -39,6 +40,11 @@
#include "omap-pcm.h"
#include "../codecs/twl4030.h"
+/* TWL4030 PMBR1 Register */
+#define TWL4030_INTBR_PMBR1 0x0D
+/* TWL4030 PMBR1 Register GPIO6 mux bit */
+#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2)
+
static struct snd_soc_card snd_soc_sdp3430;
static int sdp3430_hw_params(struct snd_pcm_substream *substream,
@@ -96,7 +102,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBS_CFM);
+ SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return ret;
@@ -280,6 +286,7 @@ static struct snd_soc_card snd_soc_sdp3430 = {
static struct twl4030_setup_data twl4030_setup = {
.ramp_delay_value = 3,
.sysclk = 26000,
+ .hs_extmute = 1,
};
/* Audio subsystem */
@@ -294,6 +301,7 @@ static struct platform_device *sdp3430_snd_device;
static int __init sdp3430_soc_init(void)
{
int ret;
+ u8 pin_mux;
if (!machine_is_omap_3430sdp()) {
pr_debug("Not SDP3430!\n");
@@ -312,6 +320,14 @@ static int __init sdp3430_soc_init(void)
*(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+ /* Set TWL4030 GPIO6 as EXTMUTE signal */
+ twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+ TWL4030_INTBR_PMBR1);
+ pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
+ pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
+ twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+ TWL4030_INTBR_PMBR1);
+
ret = platform_device_add(sdp3430_snd_device);
if (ret)
goto err1;
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
new file mode 100644
index 0000000..f90b45f
--- /dev/null
+++ b/sound/soc/omap/zoom2.c
@@ -0,0 +1,314 @@
+/*
+ * zoom2.c -- SoC audio for Zoom2
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
+#define ZOOM2_HEADSET_EXTMUTE_GPIO 153
+
+static int zoom2_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops zoom2_ops = {
+ .hw_params = zoom2_hw_params,
+};
+
+static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops zoom2_voice_ops = {
+ .hw_params = zoom2_hw_voice_params,
+};
+
+/* Zoom2 machine DAPM */
+static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Ext Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* External Mics: MAINMIC, SUBMIC with bias*/
+ {"MAINMIC", NULL, "Mic Bias 1"},
+ {"SUBMIC", NULL, "Mic Bias 2"},
+ {"Mic Bias 1", NULL, "Ext Mic"},
+ {"Mic Bias 2", NULL, "Ext Mic"},
+
+ /* External Speakers: HFL, HFR */
+ {"Ext Spk", NULL, "HFL"},
+ {"Ext Spk", NULL, "HFR"},
+
+ /* Headset Stereophone: HSOL, HSOR */
+ {"Headset Stereophone", NULL, "HSOL"},
+ {"Headset Stereophone", NULL, "HSOR"},
+
+ /* Headset Mic: HSMIC with bias */
+ {"HSMIC", NULL, "Headset Mic Bias"},
+ {"Headset Mic Bias", NULL, "Headset Mic"},
+
+ /* Aux In: AUXL, AUXR */
+ {"Aux In", NULL, "AUXL"},
+ {"Aux In", NULL, "AUXR"},
+};
+
+static int zoom2_twl4030_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ /* Add Zoom2 specific widgets */
+ ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+ ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
+ if (ret)
+ return ret;
+
+ /* Set up Zoom2 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* Zoom2 connected pins */
+ snd_soc_dapm_enable_pin(codec, "Ext Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(codec, "Aux In");
+
+ /* TWL4030 not connected pins */
+ snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+ snd_soc_dapm_nc_pin(codec, "OUTL");
+ snd_soc_dapm_nc_pin(codec, "OUTR");
+ snd_soc_dapm_nc_pin(codec, "EARPIECE");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+ snd_soc_dapm_nc_pin(codec, "CARKITL");
+ snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+ ret = snd_soc_dapm_sync(codec);
+
+ return ret;
+}
+
+static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+ unsigned short reg;
+
+ /* Enable voice interface */
+ reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+ reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+ codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link zoom2_dai[] = {
+ {
+ .name = "TWL4030 I2S",
+ .stream_name = "TWL4030 Audio",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .init = zoom2_twl4030_init,
+ .ops = &zoom2_ops,
+ },
+ {
+ .name = "TWL4030 PCM",
+ .stream_name = "TWL4030 Voice",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+ .init = zoom2_twl4030_voice_init,
+ .ops = &zoom2_voice_ops,
+ },
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_zoom2 = {
+ .name = "Zoom2",
+ .platform = &omap_soc_platform,
+ .dai_link = zoom2_dai,
+ .num_links = ARRAY_SIZE(zoom2_dai),
+};
+
+/* EXTMUTE callback function */
+void zoom2_set_hs_extmute(int mute)
+{
+ gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute);
+}
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+ .ramp_delay_value = 3, /* 161 ms */
+ .sysclk = 26000,
+ .hs_extmute = 1,
+ .set_hs_extmute = zoom2_set_hs_extmute,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device zoom2_snd_devdata = {
+ .card = &snd_soc_zoom2,
+ .codec_dev = &soc_codec_dev_twl4030,
+ .codec_data = &twl4030_setup,
+};
+
+static struct platform_device *zoom2_snd_device;
+
+static int __init zoom2_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap_zoom2()) {
+ pr_debug("Not Zoom2!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "Zoom2 SoC init\n");
+
+ zoom2_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!zoom2_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata);
+ zoom2_snd_devdata.dev = &zoom2_snd_device->dev;
+ *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+
+ ret = platform_device_add(zoom2_snd_device);
+ if (ret)
+ goto err1;
+
+ BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
+ gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
+
+ BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
+ gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(zoom2_snd_device);
+
+ return ret;
+}
+module_init(zoom2_soc_init);
+
+static void __exit zoom2_soc_exit(void)
+{
+ gpio_free(ZOOM2_HEADSET_MUX_GPIO);
+ gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
+
+ platform_device_unregister(zoom2_snd_device);
+}
+module_exit(zoom2_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC Zoom2");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 326955d..9f7c61e 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -20,12 +20,14 @@
#include <linux/platform_device.h>
#include <linux/delay.h>
#include <linux/gpio.h>
+#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include <sound/uda1380.h>
#include <mach/magician.h>
#include <asm/mach-types.h>
@@ -188,7 +190,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
if (ret < 0)
return ret;
@@ -447,34 +449,47 @@ static struct snd_soc_card snd_soc_card_magician = {
.platform = &pxa2xx_soc_platform,
};
-/* magician audio private data */
-static struct uda1380_setup_data magician_uda1380_setup = {
- .i2c_address = 0x18,
- .dac_clk = UDA1380_DAC_CLK_WSPLL,
-};
-
/* magician audio subsystem */
static struct snd_soc_device magician_snd_devdata = {
.card = &snd_soc_card_magician,
.codec_dev = &soc_codec_dev_uda1380,
- .codec_data = &magician_uda1380_setup,
};
static struct platform_device *magician_snd_device;
+/*
+ * FIXME: move into magician board file once merged into the pxa tree
+ */
+static struct uda1380_platform_data uda1380_info = {
+ .gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
+ .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
+ .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+static struct i2c_board_info i2c_board_info[] = {
+ {
+ I2C_BOARD_INFO("uda1380", 0x18),
+ .platform_data = &uda1380_info,
+ },
+};
+
static int __init magician_init(void)
{
int ret;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
if (!machine_is_magician())
return -ENODEV;
- ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
- if (ret)
- goto err_request_power;
- ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
- if (ret)
- goto err_request_reset;
+ adapter = i2c_get_adapter(0);
+ if (!adapter)
+ return -ENODEV;
+ client = i2c_new_device(adapter, i2c_board_info);
+ i2c_put_adapter(adapter);
+ if (!client)
+ return -ENODEV;
+
ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
if (ret)
goto err_request_spk;
@@ -491,14 +506,8 @@ static int __init magician_init(void)
if (ret)
goto err_request_in_sel1;
- gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
- /* we may need to have the clock running here - pH5 */
- gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
- udelay(5);
- gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
-
magician_snd_device = platform_device_alloc("soc-audio", -1);
if (!magician_snd_device) {
ret = -ENOMEM;
@@ -526,10 +535,6 @@ err_request_mic:
err_request_ep:
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
err_request_spk:
- gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
-err_request_reset:
- gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
-err_request_power:
return ret;
}
@@ -540,15 +545,12 @@ static void __exit magician_exit(void)
gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
- gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
gpio_free(EGPIO_MAGICIAN_IN_SEL1);
gpio_free(EGPIO_MAGICIAN_IN_SEL0);
gpio_free(EGPIO_MAGICIAN_MIC_POWER);
gpio_free(EGPIO_MAGICIAN_EP_POWER);
gpio_free(EGPIO_MAGICIAN_SPK_POWER);
- gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
- gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
}
module_init(magician_init);
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index e6102fd..1f96e32 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -17,13 +17,12 @@
#include <linux/moduleparam.h>
#include <linux/device.h>
#include <linux/gpio.h>
-#include <linux/interrupt.h>
-#include <linux/irq.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
+#include <sound/jack.h>
#include <asm/mach-types.h>
#include <mach/audio.h>
@@ -33,90 +32,31 @@
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
-static int palm27x_jack_func = 1;
-static int palm27x_spk_func = 1;
-static int palm27x_ep_gpio = -1;
+static struct snd_soc_jack hs_jack;
-static void palm27x_ext_control(struct snd_soc_codec *codec)
-{
- if (!palm27x_spk_func)
- snd_soc_dapm_enable_pin(codec, "Speaker");
- else
- snd_soc_dapm_disable_pin(codec, "Speaker");
-
- if (!palm27x_jack_func)
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- else
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-
- snd_soc_dapm_sync(codec);
-}
-
-static int palm27x_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->socdev->card->codec;
-
- /* check the jack status at stream startup */
- palm27x_ext_control(codec);
- return 0;
-}
-
-static struct snd_soc_ops palm27x_ops = {
- .startup = palm27x_startup,
+/* Headphones jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
};
-static irqreturn_t palm27x_interrupt(int irq, void *v)
-{
- palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
- palm27x_jack_func = !palm27x_spk_func;
- return IRQ_HANDLED;
-}
-
-static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = palm27x_jack_func;
- return 0;
-}
-
-static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
- if (palm27x_jack_func == ucontrol->value.integer.value[0])
- return 0;
-
- palm27x_jack_func = ucontrol->value.integer.value[0];
- palm27x_ext_control(codec);
- return 1;
-}
-
-static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = palm27x_spk_func;
- return 0;
-}
-
-static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
- if (palm27x_spk_func == ucontrol->value.integer.value[0])
- return 0;
-
- palm27x_spk_func = ucontrol->value.integer.value[0];
- palm27x_ext_control(codec);
- return 1;
-}
+/* Headphones jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ [0] = {
+ /* gpio is set on per-platform basis */
+ .name = "hp-gpio",
+ .report = SND_JACK_HEADPHONE,
+ .debounce_time = 200,
+ },
+};
-/* PalmTX machine dapm widgets */
+/* Palm27x machine dapm widgets */
static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_SPK("Ext. Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext. Microphone", NULL),
};
/* PalmTX audio map */
@@ -126,46 +66,66 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "HPOUTR"},
/* ext speaker connected to ROUT2, LOUT2 */
- {"Speaker", NULL, "LOUT2"},
- {"Speaker", NULL, "ROUT2"},
-};
+ {"Ext. Speaker", NULL, "LOUT2"},
+ {"Ext. Speaker", NULL, "ROUT2"},
-static const char *jack_function[] = {"Headphone", "Off"};
-static const char *spk_function[] = {"On", "Off"};
-static const struct soc_enum palm27x_enum[] = {
- SOC_ENUM_SINGLE_EXT(2, jack_function),
- SOC_ENUM_SINGLE_EXT(2, spk_function),
+ /* mic connected to MIC1 */
+ {"Ext. Microphone", NULL, "MIC1"},
};
-static const struct snd_kcontrol_new palm27x_controls[] = {
- SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
- palm27x_set_jack),
- SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
- palm27x_set_spk),
-};
+static struct snd_soc_card palm27x_asoc;
static int palm27x_ac97_init(struct snd_soc_codec *codec)
{
int err;
+ /* add palm27x specific widgets */
+ err = snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+ ARRAY_SIZE(palm27x_dapm_widgets));
+ if (err)
+ return err;
+
+ /* set up palm27x specific audio path audio_map */
+ err = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ if (err)
+ return err;
+
+ /* connected pins */
+ if (machine_is_palmld())
+ snd_soc_dapm_enable_pin(codec, "MIC1");
+ snd_soc_dapm_enable_pin(codec, "HPOUTL");
+ snd_soc_dapm_enable_pin(codec, "HPOUTR");
+ snd_soc_dapm_enable_pin(codec, "LOUT2");
+ snd_soc_dapm_enable_pin(codec, "ROUT2");
+
+ /* not connected pins */
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "MONOOUT");
+ snd_soc_dapm_nc_pin(codec, "LINEINL");
+ snd_soc_dapm_nc_pin(codec, "LINEINR");
+ snd_soc_dapm_nc_pin(codec, "PCBEEP");
+ snd_soc_dapm_nc_pin(codec, "PHONE");
+ snd_soc_dapm_nc_pin(codec, "MIC2");
+
+ err = snd_soc_dapm_sync(codec);
+ if (err)
+ return err;
- /* add palm27x specific controls */
- err = snd_soc_add_controls(codec, palm27x_controls,
- ARRAY_SIZE(palm27x_controls));
- if (err < 0)
+ /* Jack detection API stuff */
+ err = snd_soc_jack_new(&palm27x_asoc, "Headphone Jack",
+ SND_JACK_HEADPHONE, &hs_jack);
+ if (err)
return err;
- /* add palm27x specific widgets */
- snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
- ARRAY_SIZE(palm27x_dapm_widgets));
+ err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ if (err)
+ return err;
- /* set up palm27x specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
- snd_soc_dapm_sync(codec);
- return 0;
+ return err;
}
static struct snd_soc_dai_link palm27x_dai[] = {
@@ -175,14 +135,12 @@ static struct snd_soc_dai_link palm27x_dai[] = {
.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
.init = palm27x_ac97_init,
- .ops = &palm27x_ops,
},
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
- .ops = &palm27x_ops,
},
};
@@ -208,27 +166,17 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
machine_is_palmld() || machine_is_palmte2()))
return -ENODEV;
- if (pdev->dev.platform_data)
- palm27x_ep_gpio = ((struct palm27x_asoc_info *)
- (pdev->dev.platform_data))->jack_gpio;
-
- ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
- if (ret)
- return ret;
- ret = gpio_direction_input(palm27x_ep_gpio);
- if (ret)
- goto err_alloc;
+ if (!pdev->dev.platform_data) {
+ dev_err(&pdev->dev, "please supply platform_data\n");
+ return -ENODEV;
+ }
- if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
- IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
- "Headphone jack", NULL))
- goto err_alloc;
+ hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *)
+ (pdev->dev.platform_data))->jack_gpio;
palm27x_snd_device = platform_device_alloc("soc-audio", -1);
- if (!palm27x_snd_device) {
- ret = -ENOMEM;
- goto err_dev;
- }
+ if (!palm27x_snd_device)
+ return -ENOMEM;
platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
@@ -241,18 +189,12 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
put_device:
platform_device_put(palm27x_snd_device);
-err_dev:
- free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
-err_alloc:
- gpio_free(palm27x_ep_gpio);
return ret;
}
static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
{
- free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
- gpio_free(palm27x_ep_gpio);
platform_device_unregister(palm27x_snd_device);
return 0;
}
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 19c4540..5b9ed64 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
* Set the active slots in TDM/Network mode
*/
static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
- unsigned int mask, int slots)
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
struct ssp_priv *priv = cpu_dai->private_data;
struct ssp_device *ssp = priv->dev.ssp;
u32 sscr0;
- sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+ sscr0 = ssp_read_reg(ssp, SSCR0);
+ sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS);
+
+ /* set slot width */
+ if (slot_width > 16)
+ sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16);
+ else
+ sscr0 |= SSCR0_DataSize(slot_width);
+
+ if (slots > 1) {
+ /* enable network mode */
+ sscr0 |= SSCR0_MOD;
- /* set number of active slots */
- sscr0 |= SSCR0_SlotsPerFrm(slots);
+ /* set number of active slots */
+ sscr0 |= SSCR0_SlotsPerFrm(slots);
+
+ /* set active slot mask */
+ ssp_write_reg(ssp, SSTSA, tx_mask);
+ ssp_write_reg(ssp, SSRSA, rx_mask);
+ }
ssp_write_reg(ssp, SSCR0, sscr0);
- /* set active slot mask */
- ssp_write_reg(ssp, SSTSA, mask);
- ssp_write_reg(ssp, SSRSA, mask);
return 0;
}
@@ -457,31 +470,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
return -EINVAL;
}
- ssp_write_reg(ssp, SSCR0, sscr0);
- ssp_write_reg(ssp, SSCR1, sscr1);
- ssp_write_reg(ssp, SSPSP, sspsp);
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspsp |= SSPSP_SFRMP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
+ break;
+ default:
+ return -EINVAL;
+ }
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
sscr0 |= SSCR0_PSP;
sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
-
/* See hw_params() */
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_NB_NF:
- sspsp |= SSPSP_SFRMP;
- break;
- case SND_SOC_DAIFMT_NB_IF:
- break;
- case SND_SOC_DAIFMT_IB_IF:
- sspsp |= SSPSP_SCMODE(2);
- break;
- case SND_SOC_DAIFMT_IB_NF:
- sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
- break;
- default:
- return -EINVAL;
- }
break;
case SND_SOC_DAIFMT_DSP_A:
@@ -489,22 +498,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
case SND_SOC_DAIFMT_DSP_B:
sscr0 |= SSCR0_MOD | SSCR0_PSP;
sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
-
- switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
- case SND_SOC_DAIFMT_NB_NF:
- sspsp |= SSPSP_SFRMP;
- break;
- case SND_SOC_DAIFMT_NB_IF:
- break;
- case SND_SOC_DAIFMT_IB_IF:
- sspsp |= SSPSP_SCMODE(2);
- break;
- case SND_SOC_DAIFMT_IB_NF:
- sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
- break;
- default:
- return -EINVAL;
- }
break;
default:
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index d9c94d7..e9ae7b3 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -22,6 +22,7 @@
#include <mach/hardware.h>
#include <mach/regs-ac97.h>
#include <mach/dma.h>
+#include <mach/audio.h>
#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
@@ -241,9 +242,18 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops);
static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
int i;
+ pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
- for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++)
+ if (pdev->id >= 0) {
+ dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
+ return -ENXIO;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) {
pxa_ac97_dai[i].dev = &pdev->dev;
+ if (pdata && pdata->codec_pdata[0])
+ pxa_ac97_dai[i].ac97_pdata = pdata->codec_pdata[0];
+ }
/* Punt most of the init to the SoC probe; we may need the machine
* driver to do interesting things with the clocking to get us up
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index df494d1..923428f 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3CXXXX chips"
- depends on ARCH_S3C2410
+ depends on ARCH_S3C2410 || ARCH_S3C64XX
+ select S3C64XX_DMA if ARCH_S3C64XX
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97 or I2S interfaces. You will also need to
@@ -38,6 +39,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
Say Y if you want to add support for SoC audio on smdk2440
with the WM8753.
+config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
+ tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
+ depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_WM8753
+ help
+ This driver provides audio support for the Openmoko Neo FreeRunner
+ smartphone.
+
config SND_S3C24XX_SOC_JIVE_WM8750
tristate "SoC I2S Audio support for Jive"
depends on SND_S3C24XX_SOC && MACH_JIVE
@@ -57,7 +67,7 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710
config SND_S3C24XX_SOC_LN2440SBC_ALC650
tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
- depends on SND_S3C24XX_SOC
+ depends on SND_S3C24XX_SOC && ARCH_S3C2410
select SND_S3C2443_SOC_AC97
select SND_SOC_AC97_CODEC
help
@@ -66,7 +76,26 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
config SND_S3C24XX_SOC_S3C24XX_UDA134X
tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
- depends on SND_S3C24XX_SOC
+ depends on SND_S3C24XX_SOC && ARCH_S3C2410
select SND_S3C24XX_SOC_I2S
select SND_SOC_L3
select SND_SOC_UDA134X
+
+config SND_S3C24XX_SOC_SIMTEC
+ tristate
+ help
+ Internal node for common S3C24XX/Simtec suppor
+
+config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23
+ tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
+ depends on SND_S3C24XX_SOC && ARCH_S3C2410
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_TLV320AIC23
+ select SND_S3C24XX_SOC_SIMTEC
+
+config SND_S3C24XX_SOC_SIMTEC_HERMES
+ tristate "SoC I2S Audio support for Simtec Hermes board"
+ depends on SND_S3C24XX_SOC && ARCH_S3C2410
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ select SND_S3C24XX_SOC_SIMTEC
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 07a93a2..99f5a7d 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -16,12 +16,21 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
# S3C24XX Machine Support
snd-soc-jive-wm8750-objs := jive_wm8750.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
+snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
+snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
+snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
new file mode 100644
index 0000000..0c52e36
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -0,0 +1,498 @@
+/*
+ * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02)
+ *
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory <linux@wolfsonmicro.com>
+ * Copyright 2009 Wolfson Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/regs-clock.h>
+#include <asm/io.h>
+#include <mach/gta02.h>
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+static struct snd_soc_card neo1973_gta02;
+
+static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0, bclk = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ pll_out = 12288000;
+ break;
+ case 48000:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 12288000;
+ break;
+ case 96000:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 12288000;
+ break;
+ case 11025:
+ bclk = WM8753_BCLK_DIV_16;
+ pll_out = 11289600;
+ break;
+ case 22050:
+ bclk = WM8753_BCLK_DIV_8;
+ pll_out = 11289600;
+ break;
+ case 44100:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 11289600;
+ break;
+ case 88200:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ /* set codec BCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai,
+ WM8753_BCLKDIV, bclk);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(4, 4));
+ if (ret < 0)
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ iis_clkrate / 4, pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_gta02_hifi_ops = {
+ .hw_params = neo1973_gta02_hifi_hw_params,
+ .hw_free = neo1973_gta02_hifi_hw_free,
+};
+
+static int neo1973_gta02_voice_hw_params(
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int pcmdiv = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ if (params_rate(params) != 8000)
+ return -EINVAL;
+ if (params_channels(params) != 1)
+ return -EINVAL;
+
+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+ /* todo: gg check mode (DSP_B) against CSR datasheet */
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
+ 12288000, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set codec PCM division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
+ pcmdiv);
+ if (ret < 0)
+ return ret;
+
+ /* configue and enable PLL for 12.288MHz output */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+ iis_clkrate / 4, 12288000);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_gta02_voice_ops = {
+ .hw_params = neo1973_gta02_voice_hw_params,
+ .hw_free = neo1973_gta02_voice_hw_free,
+};
+
+#define LM4853_AMP 1
+#define LM4853_SPK 2
+
+static u8 lm4853_state;
+
+/* This has no effect, it exists only to maintain compatibility with
+ * existing ALSA state files.
+ */
+static int lm4853_set_state(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int val = ucontrol->value.integer.value[0];
+
+ if (val)
+ lm4853_state |= LM4853_AMP;
+ else
+ lm4853_state &= ~LM4853_AMP;
+
+ return 0;
+}
+
+static int lm4853_get_state(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
+
+ return 0;
+}
+
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int val = ucontrol->value.integer.value[0];
+
+ if (val) {
+ lm4853_state |= LM4853_SPK;
+ gpio_set_value(GTA02_GPIO_HP_IN, 0);
+ } else {
+ lm4853_state &= ~LM4853_SPK;
+ gpio_set_value(GTA02_GPIO_HP_IN, 1);
+ }
+
+ return 0;
+}
+
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
+
+ return 0;
+}
+
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k,
+ int event)
+{
+ gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value));
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* Connections to the lm4853 amp */
+ {"Stereo Out", NULL, "LOUT1"},
+ {"Stereo Out", NULL, "ROUT1"},
+
+ /* Connections to the GSM Module */
+ {"GSM Line Out", NULL, "MONO1"},
+ {"GSM Line Out", NULL, "MONO2"},
+ {"RXP", NULL, "GSM Line In"},
+ {"RXN", NULL, "GSM Line In"},
+
+ /* Connections to Headset */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Call Mic */
+ {"MIC2", NULL, "Mic Bias"},
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Handset Mic"},
+
+ /* Call Speaker */
+ {"Handset Spk", NULL, "LOUT2"},
+ {"Handset Spk", NULL, "ROUT2"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Stereo Out"),
+ SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+ SOC_DAPM_PIN_SWITCH("GSM Line In"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Handset Mic"),
+ SOC_DAPM_PIN_SWITCH("Handset Spk"),
+
+ /* This has no effect, it exists only to maintain compatibility with
+ * existing ALSA state files.
+ */
+ SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
+ lm4853_get_state,
+ lm4853_set_state),
+ SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
+ lm4853_get_spk,
+ lm4853_set_spk),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 GTA02.
+ */
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+ int err;
+
+ /* set up NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "OUT4");
+ snd_soc_dapm_nc_pin(codec, "LINE1");
+ snd_soc_dapm_nc_pin(codec, "LINE2");
+
+ /* Add neo1973 gta02 specific widgets */
+ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
+
+ /* add neo1973 gta02 specific controls */
+ err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
+ ARRAY_SIZE(wm8753_neo1973_gta02_controls));
+
+ if (err < 0)
+ return err;
+
+ /* set up neo1973 gta02 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* set endpoints to default off mode */
+ snd_soc_dapm_disable_pin(codec, "Stereo Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Handset Mic");
+ snd_soc_dapm_disable_pin(codec, "Handset Spk");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_dai bt_dai = {
+ .name = "Bluetooth",
+ .id = 0,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+};
+
+static struct snd_soc_dai_link neo1973_gta02_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+ .name = "WM8753",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+ .init = neo1973_gta02_wm8753_init,
+ .ops = &neo1973_gta02_hifi_ops,
+},
+{ /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+ .ops = &neo1973_gta02_voice_ops,
+},
+};
+
+static struct snd_soc_card neo1973_gta02 = {
+ .name = "neo1973-gta02",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = neo1973_gta02_dai,
+ .num_links = ARRAY_SIZE(neo1973_gta02_dai),
+};
+
+static struct snd_soc_device neo1973_gta02_snd_devdata = {
+ .card = &neo1973_gta02,
+ .codec_dev = &soc_codec_dev_wm8753,
+};
+
+static struct platform_device *neo1973_gta02_snd_device;
+
+static int __init neo1973_gta02_init(void)
+{
+ int ret;
+
+ if (!machine_is_neo1973_gta02()) {
+ printk(KERN_INFO
+ "Only GTA02 is supported by this ASoC driver\n");
+ return -ENODEV;
+ }
+
+ /* register bluetooth DAI here */
+ ret = snd_soc_register_dai(&bt_dai);
+ if (ret)
+ return ret;
+
+ neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!neo1973_gta02_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(neo1973_gta02_snd_device,
+ &neo1973_gta02_snd_devdata);
+ neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
+ ret = platform_device_add(neo1973_gta02_snd_device);
+
+ if (ret) {
+ platform_device_put(neo1973_gta02_snd_device);
+ return ret;
+ }
+
+ /* Initialise GPIOs used by amp */
+ ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
+ if (ret) {
+ pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
+ goto err_unregister_device;
+ }
+
+ ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1);
+ if (ret) {
+ pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
+ goto err_free_gpio_hp_in;
+ }
+
+ ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
+ if (ret) {
+ pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
+ goto err_free_gpio_hp_in;
+ }
+
+ ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
+ if (ret) {
+ pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
+ goto err_free_gpio_amp_shut;
+ }
+
+ return 0;
+
+err_free_gpio_amp_shut:
+ gpio_free(GTA02_GPIO_AMP_SHUT);
+err_free_gpio_hp_in:
+ gpio_free(GTA02_GPIO_HP_IN);
+err_unregister_device:
+ platform_device_unregister(neo1973_gta02_snd_device);
+ return ret;
+}
+module_init(neo1973_gta02_init);
+
+static void __exit neo1973_gta02_exit(void)
+{
+ snd_soc_unregister_dai(&bt_dai);
+ platform_device_unregister(neo1973_gta02_snd_device);
+ gpio_free(GTA02_GPIO_HP_IN);
+ gpio_free(GTA02_GPIO_AMP_SHUT);
+}
+module_exit(neo1973_gta02_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 1a28317..aa7af0b 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -36,6 +36,7 @@
#include <mach/dma.h>
#include "s3c-i2s-v2.h"
+#include "s3c24xx-pcm.h"
#undef S3C_IIS_V2_SUPPORTED
@@ -357,19 +358,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
#endif
#ifdef CONFIG_PLAT_S3C64XX
- iismod &= ~0x606;
+ iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK);
/* Sample size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
/* 8 bit sample, 16fs BCLK */
- iismod |= 0x2004;
+ iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS);
break;
case SNDRV_PCM_FORMAT_S16_LE:
/* 16 bit sample, 32fs BCLK */
break;
case SNDRV_PCM_FORMAT_S24_LE:
/* 24 bit sample, 48fs BCLK */
- iismod |= 0x4002;
+ iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS);
break;
}
#endif
@@ -387,6 +388,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
int ret = 0;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
pr_debug("Entered %s\n", __func__);
@@ -416,6 +419,14 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
s3c2412_snd_txctrl(i2s, 1);
local_irq_restore(irqs);
+
+ /*
+ * Load the next buffer to DMA to meet the reqirement
+ * of the auto reload mechanism of S3C24XX.
+ * This call won't bother S3C64XX.
+ */
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
break;
case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 3f03d5d..fc1beb0 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97;
static DECLARE_COMPLETION(ac97_completion);
static u32 codec_ready;
-static DECLARE_MUTEX(ac97_mutex);
+static DEFINE_MUTEX(ac97_mutex);
static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
unsigned short reg)
@@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
u32 ac_codec_cmd;
u32 stat, addr, data;
- down(&ac97_mutex);
+ mutex_lock(&ac97_mutex);
codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
" rep addr = %02x\n", reg, addr);
- up(&ac97_mutex);
+ mutex_unlock(&ac97_mutex);
return (unsigned short)data;
}
@@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
u32 ac_glbctrl;
u32 ac_codec_cmd;
- down(&ac97_mutex);
+ mutex_lock(&ac97_mutex);
codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
- up(&ac97_mutex);
+ mutex_unlock(&ac97_mutex);
}
@@ -290,6 +290,9 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
switch (cmd) {
@@ -312,6 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
}
writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
return 0;
}
@@ -334,6 +339,9 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
u32 ac_glbctrl;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
switch (cmd) {
@@ -349,6 +357,8 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
}
writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
return 0;
}
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 556e35f..40e2c47 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -279,6 +279,9 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int channel = ((struct s3c24xx_pcm_dma_params *)
+ rtd->dai->cpu_dai->dma_data)->channel;
pr_debug("Entered %s\n", __func__);
@@ -296,6 +299,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
s3c24xx_snd_rxctrl(1);
else
s3c24xx_snd_txctrl(1);
+
+ s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index eecfa5e..5cbbdc8 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -255,7 +255,6 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
- s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
@@ -318,6 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
pr_debug("Entered %s\n", __func__);
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
new file mode 100644
index 0000000..1966e0d
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -0,0 +1,394 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+static struct s3c24xx_audio_simtec_pdata *pdata;
+static struct clk *xtal_clk;
+
+static int spk_gain;
+static int spk_unmute;
+
+/**
+ * speaker_gain_get - read the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spk_gain;
+ return 0;
+}
+
+/**
+ * speaker_gain_set - set the value of the speaker amp gain
+ * @value: The value to write.
+ */
+static void speaker_gain_set(int value)
+{
+ gpio_set_value_cansleep(pdata->amp_gain[0], value & 1);
+ gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1);
+}
+
+/**
+ * speaker_gain_put - set the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ *
+ * Note, if the speaker amp is muted, then we do not set a gain value
+ * as at-least one of the ICs that is fitted will try and power up even
+ * if the main control is set to off.
+ */
+static int speaker_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int value = ucontrol->value.integer.value[0];
+
+ spk_gain = value;
+
+ if (!spk_unmute)
+ speaker_gain_set(value);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new amp_gain_controls[] = {
+ SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0,
+ speaker_gain_get, speaker_gain_put),
+};
+
+/**
+ * spk_unmute_state - set the unmute state of the speaker
+ * @to: zero to unmute, non-zero to ununmute.
+ */
+static void spk_unmute_state(int to)
+{
+ pr_debug("%s: to=%d\n", __func__, to);
+
+ spk_unmute = to;
+ gpio_set_value(pdata->amp_gpio, to);
+
+ /* if we're umuting, also re-set the gain */
+ if (to && pdata->amp_gain[0] > 0)
+ speaker_gain_set(spk_gain);
+}
+
+/**
+ * speaker_unmute_get - read the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_unmute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spk_unmute;
+ return 0;
+}
+
+/**
+ * speaker_unmute_put - set the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ */
+static int speaker_unmute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ spk_unmute_state(ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+/* This is added as a manual control as the speaker amps create clicks
+ * when their power state is changed, which are far more noticeable than
+ * anything produced by the CODEC itself.
+ */
+static const struct snd_kcontrol_new amp_unmute_controls[] = {
+ SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0,
+ speaker_unmute_get, speaker_unmute_put),
+};
+
+void simtec_audio_init(struct snd_soc_codec *codec)
+{
+ if (pdata->amp_gpio > 0) {
+ pr_debug("%s: adding amp routes\n", __func__);
+
+ snd_soc_add_controls(codec, amp_unmute_controls,
+ ARRAY_SIZE(amp_unmute_controls));
+ }
+
+ if (pdata->amp_gain[0] > 0) {
+ pr_debug("%s: adding amp controls\n", __func__);
+ snd_soc_add_controls(codec, amp_gain_controls,
+ ARRAY_SIZE(amp_gain_controls));
+ }
+}
+EXPORT_SYMBOL_GPL(simtec_audio_init);
+
+#define CODEC_CLOCK 12000000
+
+/**
+ * simtec_hw_params - update hardware parameters
+ * @substream: The audio substream instance.
+ * @params: The parameters requested.
+ *
+ * Update the codec data routing and configuration settings
+ * from the supplied data.
+ */
+static int simtec_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set the CODEC as the bus clock master, I2S */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ pr_err("%s: failed set cpu dai format\n", __func__);
+ return ret;
+ }
+
+ /* Set the CODEC as the bus clock master */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ pr_err("%s: failed set codec dai format\n", __func__);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ CODEC_CLOCK, SND_SOC_CLOCK_IN);
+ if (ret) {
+ pr_err( "%s: failed setting codec sysclk\n", __func__);
+ return ret;
+ }
+
+ if (pdata->use_mpllin) {
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL,
+ 0, SND_SOC_CLOCK_OUT);
+
+ if (ret) {
+ pr_err("%s: failed to set MPLLin as clksrc\n",
+ __func__);
+ return ret;
+ }
+ }
+
+ if (pdata->output_cdclk) {
+ int cdclk_scale;
+
+ cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK;
+ cdclk_scale--;
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ cdclk_scale);
+ }
+
+ return 0;
+}
+
+static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd)
+{
+ /* call any board supplied startup code, this currently only
+ * covers the bast/vr1000 which have a CPLD in the way of the
+ * LRCLK */
+ if (pd->startup)
+ pd->startup();
+
+ return 0;
+}
+
+static struct snd_soc_ops simtec_snd_ops = {
+ .hw_params = simtec_hw_params,
+};
+
+/**
+ * attach_gpio_amp - get and configure the necessary gpios
+ * @dev: The device we're probing.
+ * @pd: The platform data supplied by the board.
+ *
+ * If there is a GPIO based amplifier attached to the board, claim
+ * the necessary GPIO lines for it, and set default values.
+ */
+static int attach_gpio_amp(struct device *dev,
+ struct s3c24xx_audio_simtec_pdata *pd)
+{
+ int ret;
+
+ /* attach gpio amp gain (if any) */
+ if (pdata->amp_gain[0] > 0) {
+ ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio gain0\n");
+ return ret;
+ }
+
+ ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio gain1\n");
+ gpio_free(pdata->amp_gain[0]);
+ return ret;
+ }
+
+ gpio_direction_output(pd->amp_gain[0], 0);
+ gpio_direction_output(pd->amp_gain[1], 0);
+ }
+
+ /* note, curently we assume GPA0 isn't valid amp */
+ if (pdata->amp_gpio > 0) {
+ ret = gpio_request(pd->amp_gpio, "gpio-amp");
+ if (ret) {
+ dev_err(dev, "cannot get amp gpio %d (%d)\n",
+ pd->amp_gpio, ret);
+ goto err_amp;
+ }
+
+ /* set the amp off at startup */
+ spk_unmute_state(0);
+ }
+
+ return 0;
+
+err_amp:
+ if (pd->amp_gain[0] > 0) {
+ gpio_free(pd->amp_gain[0]);
+ gpio_free(pd->amp_gain[1]);
+ }
+
+ return ret;
+}
+
+static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd)
+{
+ if (pd->amp_gain[0] > 0) {
+ gpio_free(pd->amp_gain[0]);
+ gpio_free(pd->amp_gain[1]);
+ }
+
+ if (pd->amp_gpio > 0)
+ gpio_free(pd->amp_gpio);
+}
+
+#ifdef CONFIG_PM
+int simtec_audio_resume(struct device *dev)
+{
+ simtec_call_startup(pdata);
+ return 0;
+}
+
+struct dev_pm_ops simtec_audio_pmops = {
+ .resume = simtec_audio_resume,
+};
+EXPORT_SYMBOL_GPL(simtec_audio_pmops);
+#endif
+
+int __devinit simtec_audio_core_probe(struct platform_device *pdev,
+ struct snd_soc_device *socdev)
+{
+ struct platform_device *snd_dev;
+ int ret;
+
+ socdev->card->dai_link->ops = &simtec_snd_ops;
+
+ pdata = pdev->dev.platform_data;
+ if (!pdata) {
+ dev_err(&pdev->dev, "no platform data supplied\n");
+ return -EINVAL;
+ }
+
+ simtec_call_startup(pdata);
+
+ xtal_clk = clk_get(&pdev->dev, "xtal");
+ if (IS_ERR(xtal_clk)) {
+ dev_err(&pdev->dev, "could not get clkout0\n");
+ return -EINVAL;
+ }
+
+ dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk));
+
+ ret = attach_gpio_amp(&pdev->dev, pdata);
+ if (ret)
+ goto err_clk;
+
+ snd_dev = platform_device_alloc("soc-audio", -1);
+ if (!snd_dev) {
+ dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n");
+ ret = -ENOMEM;
+ goto err_gpio;
+ }
+
+ platform_set_drvdata(snd_dev, socdev);
+ socdev->dev = &snd_dev->dev;
+
+ ret = platform_device_add(snd_dev);
+ if (ret) {
+ dev_err(&pdev->dev, "failed to add soc-audio dev\n");
+ goto err_pdev;
+ }
+
+ platform_set_drvdata(pdev, snd_dev);
+ return 0;
+
+err_pdev:
+ platform_device_put(snd_dev);
+
+err_gpio:
+ detach_gpio_amp(pdata);
+
+err_clk:
+ clk_put(xtal_clk);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_core_probe);
+
+int __devexit simtec_audio_remove(struct platform_device *pdev)
+{
+ struct platform_device *snd_dev = platform_get_drvdata(pdev);
+
+ platform_device_unregister(snd_dev);
+
+ detach_gpio_amp(pdata);
+ clk_put(xtal_clk);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_remove);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
new file mode 100644
index 0000000..2714203
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -0,0 +1,22 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.h
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+extern void simtec_audio_init(struct snd_soc_codec *codec);
+
+extern int simtec_audio_core_probe(struct platform_device *pdev,
+ struct snd_soc_device *socdev);
+
+extern int simtec_audio_remove(struct platform_device *pdev);
+
+#ifdef CONFIG_PM
+extern struct dev_pm_ops simtec_audio_pmops;
+#define simtec_audio_pm &simtec_audio_pmops
+#else
+#define simtec_audio_pm NULL
+#endif
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
new file mode 100644
index 0000000..8346bd9
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -0,0 +1,153 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic3x.h"
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("GSM Out", NULL),
+ SND_SOC_DAPM_LINE("GSM In", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_LINE("ZV", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+ /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */
+
+ { "Headphone Jack", NULL, "HPLOUT" },
+ { "Headphone Jack", NULL, "HPLCOM" },
+ { "Headphone Jack", NULL, "HPROUT" },
+ { "Headphone Jack", NULL, "HPRCOM" },
+
+ /* ZV connected to Line1 */
+
+ { "LINE1L", NULL, "ZV" },
+ { "LINE1R", NULL, "ZV" },
+
+ /* Line In connected to Line2 */
+
+ { "LINE2L", NULL, "Line In" },
+ { "LINE2R", NULL, "Line In" },
+
+ /* Microphone connected to MIC3R and MIC_BIAS */
+
+ { "MIC3L", NULL, "Mic Jack" },
+
+ /* GSM connected to MONO_LOUT and MIC3L (in) */
+
+ { "GSM Out", NULL, "MONO_LOUT" },
+ { "MIC3L", NULL, "GSM In" },
+
+ /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are
+ * not using the DAPM to power it up and down as there it makes
+ * a click when powering up. */
+};
+
+/**
+ * simtec_hermes_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_hermes_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, dapm_widgets,
+ ARRAY_SIZE(dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ simtec_audio_init(codec);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct aic3x_setup_data codec_setup = {
+};
+
+static struct snd_soc_dai_link simtec_dai_aic33 = {
+ .name = "tlv320aic33",
+ .stream_name = "TLV320AIC33",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &aic3x_dai,
+ .init = simtec_hermes_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic33 = {
+ .name = "Simtec-Hermes",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = &simtec_dai_aic33,
+ .num_links = 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic33 = {
+ .card = &snd_soc_machine_simtec_aic33,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &codec_setup,
+};
+
+static int __devinit simtec_audio_hermes_probe(struct platform_device *pd)
+{
+ dev_info(&pd->dev, "probing....\n");
+ return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33);
+}
+
+static struct platform_driver simtec_audio_hermes_platdrv = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "s3c24xx-simtec-hermes-snd",
+ .pm = simtec_audio_pm,
+ },
+ .probe = simtec_audio_hermes_probe,
+ .remove = __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
+
+static int __init simtec_hermes_modinit(void)
+{
+ return platform_driver_register(&simtec_audio_hermes_platdrv);
+}
+
+static void __exit simtec_hermes_modexit(void)
+{
+ platform_driver_unregister(&simtec_audio_hermes_platdrv);
+}
+
+module_init(simtec_hermes_modinit);
+module_exit(simtec_hermes_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
new file mode 100644
index 0000000..25797e0
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -0,0 +1,137 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic23.h"
+
+/* supported machines:
+ *
+ * Machine Connections AMP
+ * ------- ----------- ---
+ * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired)
+ * VR1000 HPOUT, LIN None
+ * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
+ * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
+ * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R)
+ */
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+ { "Headphone Jack", NULL, "LHPOUT"},
+ { "Headphone Jack", NULL, "RHPOUT"},
+
+ { "Line Out", NULL, "LOUT" },
+ { "Line Out", NULL, "ROUT" },
+
+ { "LLINEIN", NULL, "Line In"},
+ { "RLINEIN", NULL, "Line In"},
+
+ { "MICIN", NULL, "Mic Jack"},
+};
+
+/**
+ * simtec_tlv320aic23_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, dapm_widgets,
+ ARRAY_SIZE(dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ simtec_audio_init(codec);
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link simtec_dai_aic23 = {
+ .name = "tlv320aic23",
+ .stream_name = "TLV320AIC23",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &tlv320aic23_dai,
+ .init = simtec_tlv320aic23_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic23 = {
+ .name = "Simtec",
+ .platform = &s3c24xx_soc_platform,
+ .dai_link = &simtec_dai_aic23,
+ .num_links = 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic23 = {
+ .card = &snd_soc_machine_simtec_aic23,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd)
+{
+ return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23);
+}
+
+static struct platform_driver simtec_audio_tlv320aic23_platdrv = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "s3c24xx-simtec-tlv320aic23",
+ .pm = simtec_audio_pm,
+ },
+ .probe = simtec_audio_tlv320aic23_probe,
+ .remove = __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23");
+
+static int __init simtec_tlv320aic23_modinit(void)
+{
+ return platform_driver_register(&simtec_audio_tlv320aic23_platdrv);
+}
+
+static void __exit simtec_tlv320aic23_modexit(void)
+{
+ platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv);
+}
+
+module_init(simtec_tlv320aic23_modinit);
+module_exit(simtec_tlv320aic23_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index b5f95f9..c1b40ac 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -14,6 +14,7 @@
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
+#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -189,8 +190,6 @@ static struct snd_soc_card snd_soc_card_s6105 = {
/* s6105 audio private data */
static struct aic3x_setup_data s6105_aic3x_setup = {
- .i2c_bus = 0,
- .i2c_address = 0x18,
};
/* s6105 audio subsystem */
@@ -211,10 +210,19 @@ static struct s6000_snd_platform_data __initdata s6105_snd_data = {
static struct platform_device *s6105_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic33", 0x18), }
+};
+
static int __init s6105_init(void)
{
int ret;
+ i2c_register_board_info(0, i2c_device, ARRAY_SIZE(i2c_device));
+
s6105_snd_device = platform_device_alloc("soc-audio", -1);
if (!s6105_snd_device)
return -ENOMEM;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 54bd604..9154b43 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -20,7 +20,12 @@ config SND_SOC_SH4_HAC
config SND_SOC_SH4_SSI
tristate
-
+config SND_SOC_SH4_FSI
+ tristate "SH4 FSI support"
+ depends on CPU_SUBTYPE_SH7724
+ select SH_DMA
+ help
+ This option enables FSI sound support
##
## Boards
@@ -35,4 +40,12 @@ config SND_SH7760_AC97
This option enables generic sound support for the first
AC97 unit of the SH7760.
+config SND_FSI_AK4642
+ bool "FSI-AK4642 sound support"
+ depends on SND_SOC_SH4_FSI
+ select SND_SOC_AK4642
+ help
+ This option enables generic sound support for the
+ FSI - AK4642 unit
+
endmenu
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
index a8e8ab8..a699787 100644
--- a/sound/soc/sh/Makefile
+++ b/sound/soc/sh/Makefile
@@ -5,10 +5,14 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o
## audio units found on some SH-4
snd-soc-hac-objs := hac.o
snd-soc-ssi-objs := ssi.o
+snd-soc-fsi-objs := fsi.o
obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
+obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o
## boards
snd-soc-sh7760-ac97-objs := sh7760-ac97.o
+snd-soc-fsi-ak4642-objs := fsi-ak4642.o
obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
+obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
new file mode 100644
index 0000000..c7af097
--- /dev/null
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -0,0 +1,107 @@
+/*
+ * FSI-AK464x sound support for ms7724se
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <sound/sh_fsi.h>
+#include <../sound/soc/codecs/ak4642.h>
+
+static struct snd_soc_dai_link fsi_dai_link = {
+ .name = "AK4642",
+ .stream_name = "AK4642",
+ .cpu_dai = &fsi_soc_dai[0], /* fsi */
+ .codec_dai = &ak4642_dai,
+ .ops = NULL,
+};
+
+static struct snd_soc_card fsi_soc_card = {
+ .name = "FSI",
+ .platform = &fsi_soc_platform,
+ .dai_link = &fsi_dai_link,
+ .num_links = 1,
+};
+
+static struct snd_soc_device fsi_snd_devdata = {
+ .card = &fsi_soc_card,
+ .codec_dev = &soc_codec_dev_ak4642,
+};
+
+#define AK4642_BUS 0
+#define AK4642_ADR 0x12
+static int ak4642_add_i2c_device(void)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = AK4642_ADR;
+ strlcpy(info.type, "ak4642", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(AK4642_BUS);
+ if (!adapter) {
+ printk(KERN_DEBUG "can't get i2c adapter\n");
+ return -ENODEV;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ printk(KERN_DEBUG "can't add i2c device\n");
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static struct platform_device *fsi_snd_device;
+
+static int __init fsi_ak4642_init(void)
+{
+ int ret = -ENOMEM;
+
+ ak4642_add_i2c_device();
+
+ fsi_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!fsi_snd_device)
+ goto out;
+
+ platform_set_drvdata(fsi_snd_device,
+ &fsi_snd_devdata);
+ fsi_snd_devdata.dev = &fsi_snd_device->dev;
+ ret = platform_device_add(fsi_snd_device);
+
+ if (ret)
+ platform_device_put(fsi_snd_device);
+
+out:
+ return ret;
+}
+
+static void __exit fsi_ak4642_exit(void)
+{
+ platform_device_unregister(fsi_snd_device);
+}
+
+module_init(fsi_ak4642_init);
+module_exit(fsi_ak4642_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
new file mode 100644
index 0000000..4412324
--- /dev/null
+++ b/sound/soc/sh/fsi.c
@@ -0,0 +1,1004 @@
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Based on ssi.c
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/list.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/sh_fsi.h>
+#include <asm/atomic.h>
+#include <asm/dma.h>
+#include <asm/dma-sh.h>
+
+#define DO_FMT 0x0000
+#define DOFF_CTL 0x0004
+#define DOFF_ST 0x0008
+#define DI_FMT 0x000C
+#define DIFF_CTL 0x0010
+#define DIFF_ST 0x0014
+#define CKG1 0x0018
+#define CKG2 0x001C
+#define DIDT 0x0020
+#define DODT 0x0024
+#define MUTE_ST 0x0028
+#define REG_END MUTE_ST
+
+#define INT_ST 0x0200
+#define IEMSK 0x0204
+#define IMSK 0x0208
+#define MUTE 0x020C
+#define CLK_RST 0x0210
+#define SOFT_RST 0x0214
+#define MREG_START INT_ST
+#define MREG_END SOFT_RST
+
+/* DO_FMT */
+/* DI_FMT */
+#define CR_FMT(param) ((param) << 4)
+# define CR_MONO 0x0
+# define CR_MONO_D 0x1
+# define CR_PCM 0x2
+# define CR_I2S 0x3
+# define CR_TDM 0x4
+# define CR_TDM_D 0x5
+
+/* DOFF_CTL */
+/* DIFF_CTL */
+#define IRQ_HALF 0x00100000
+#define FIFO_CLR 0x00000001
+
+/* DOFF_ST */
+#define ERR_OVER 0x00000010
+#define ERR_UNDER 0x00000001
+
+/* CLK_RST */
+#define B_CLK 0x00000010
+#define A_CLK 0x00000001
+
+/* INT_ST */
+#define INT_B_IN (1 << 12)
+#define INT_B_OUT (1 << 8)
+#define INT_A_IN (1 << 4)
+#define INT_A_OUT (1 << 0)
+
+#define FSI_RATES SNDRV_PCM_RATE_8000_96000
+
+#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+
+/************************************************************************
+
+
+ struct
+
+
+************************************************************************/
+struct fsi_priv {
+ void __iomem *base;
+ struct snd_pcm_substream *substream;
+
+ int fifo_max;
+ int chan;
+ int dma_chan;
+
+ int byte_offset;
+ int period_len;
+ int buffer_len;
+ int periods;
+};
+
+struct fsi_master {
+ void __iomem *base;
+ int irq;
+ struct clk *clk;
+ struct fsi_priv fsia;
+ struct fsi_priv fsib;
+ struct sh_fsi_platform_info *info;
+};
+
+static struct fsi_master *master;
+
+/************************************************************************
+
+
+ basic read write function
+
+
+************************************************************************/
+static int __fsi_reg_write(u32 reg, u32 data)
+{
+ /* valid data area is 24bit */
+ data &= 0x00ffffff;
+
+ return ctrl_outl(data, reg);
+}
+
+static u32 __fsi_reg_read(u32 reg)
+{
+ return ctrl_inl(reg);
+}
+
+static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data)
+{
+ u32 val = __fsi_reg_read(reg);
+
+ val &= ~mask;
+ val |= data & mask;
+
+ return __fsi_reg_write(reg, val);
+}
+
+static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data)
+{
+ if (reg > REG_END)
+ return -1;
+
+ return __fsi_reg_write((u32)(fsi->base + reg), data);
+}
+
+static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg)
+{
+ if (reg > REG_END)
+ return 0;
+
+ return __fsi_reg_read((u32)(fsi->base + reg));
+}
+
+static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data)
+{
+ if (reg > REG_END)
+ return -1;
+
+ return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data);
+}
+
+static int fsi_master_write(u32 reg, u32 data)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return -1;
+
+ return __fsi_reg_write((u32)(master->base + reg), data);
+}
+
+static u32 fsi_master_read(u32 reg)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return 0;
+
+ return __fsi_reg_read((u32)(master->base + reg));
+}
+
+static int fsi_master_mask_set(u32 reg, u32 mask, u32 data)
+{
+ if ((reg < MREG_START) ||
+ (reg > MREG_END))
+ return -1;
+
+ return __fsi_reg_mask_set((u32)(master->base + reg), mask, data);
+}
+
+/************************************************************************
+
+
+ basic function
+
+
+************************************************************************/
+static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd;
+ struct fsi_priv *fsi = NULL;
+
+ if (!substream || !master)
+ return NULL;
+
+ rtd = substream->private_data;
+ switch (rtd->dai->cpu_dai->id) {
+ case 0:
+ fsi = &master->fsia;
+ break;
+ case 1:
+ fsi = &master->fsib;
+ break;
+ }
+
+ return fsi;
+}
+
+static int fsi_is_port_a(struct fsi_priv *fsi)
+{
+ /* return
+ * 1 : port a
+ * 0 : port b
+ */
+
+ if (fsi == &master->fsia)
+ return 1;
+
+ return 0;
+}
+
+static u32 fsi_get_info_flags(struct fsi_priv *fsi)
+{
+ int is_porta = fsi_is_port_a(fsi);
+
+ return is_porta ? master->info->porta_flags :
+ master->info->portb_flags;
+}
+
+static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play)
+{
+ u32 mode;
+ u32 flags = fsi_get_info_flags(fsi);
+
+ mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE;
+
+ /* return
+ * 1 : master mode
+ * 0 : slave mode
+ */
+
+ return (mode & flags) != mode;
+}
+
+static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play)
+{
+ int is_porta = fsi_is_port_a(fsi);
+ u32 data;
+
+ if (is_porta)
+ data = is_play ? (1 << 0) : (1 << 4);
+ else
+ data = is_play ? (1 << 8) : (1 << 12);
+
+ return data;
+}
+
+static void fsi_stream_push(struct fsi_priv *fsi,
+ struct snd_pcm_substream *substream,
+ u32 buffer_len,
+ u32 period_len)
+{
+ fsi->substream = substream;
+ fsi->buffer_len = buffer_len;
+ fsi->period_len = period_len;
+ fsi->byte_offset = 0;
+ fsi->periods = 0;
+}
+
+static void fsi_stream_pop(struct fsi_priv *fsi)
+{
+ fsi->substream = NULL;
+ fsi->buffer_len = 0;
+ fsi->period_len = 0;
+ fsi->byte_offset = 0;
+ fsi->periods = 0;
+}
+
+static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play)
+{
+ u32 status;
+ u32 reg = is_play ? DOFF_ST : DIFF_ST;
+ int residue;
+
+ status = fsi_reg_read(fsi, reg);
+ residue = 0x1ff & (status >> 8);
+ residue *= fsi->chan;
+
+ return residue;
+}
+
+static int fsi_get_residue(struct fsi_priv *fsi, int is_play)
+{
+ int residue;
+ int width;
+ struct snd_pcm_runtime *runtime;
+
+ runtime = fsi->substream->runtime;
+
+ /* get 1 channel data width */
+ width = frames_to_bytes(runtime, 1) / fsi->chan;
+
+ if (2 == width)
+ residue = fsi_get_fifo_residue(fsi, is_play);
+ else
+ residue = get_dma_residue(fsi->dma_chan);
+
+ return residue;
+}
+
+/************************************************************************
+
+
+ basic dma function
+
+
+************************************************************************/
+#define PORTA_DMA 0
+#define PORTB_DMA 1
+
+static int fsi_get_dma_chan(void)
+{
+ if (0 != request_dma(PORTA_DMA, "fsia"))
+ return -EIO;
+
+ if (0 != request_dma(PORTB_DMA, "fsib")) {
+ free_dma(PORTA_DMA);
+ return -EIO;
+ }
+
+ master->fsia.dma_chan = PORTA_DMA;
+ master->fsib.dma_chan = PORTB_DMA;
+
+ return 0;
+}
+
+static void fsi_free_dma_chan(void)
+{
+ dma_wait_for_completion(PORTA_DMA);
+ dma_wait_for_completion(PORTB_DMA);
+ free_dma(PORTA_DMA);
+ free_dma(PORTB_DMA);
+
+ master->fsia.dma_chan = -1;
+ master->fsib.dma_chan = -1;
+}
+
+/************************************************************************
+
+
+ ctrl function
+
+
+************************************************************************/
+static void fsi_irq_enable(struct fsi_priv *fsi, int is_play)
+{
+ u32 data = fsi_port_ab_io_bit(fsi, is_play);
+
+ fsi_master_mask_set(IMSK, data, data);
+ fsi_master_mask_set(IEMSK, data, data);
+}
+
+static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
+{
+ u32 data = fsi_port_ab_io_bit(fsi, is_play);
+
+ fsi_master_mask_set(IMSK, data, 0);
+ fsi_master_mask_set(IEMSK, data, 0);
+}
+
+static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable)
+{
+ u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4);
+
+ if (enable)
+ fsi_master_mask_set(CLK_RST, val, val);
+ else
+ fsi_master_mask_set(CLK_RST, val, 0);
+}
+
+static void fsi_irq_init(struct fsi_priv *fsi, int is_play)
+{
+ u32 data;
+ u32 ctrl;
+
+ data = fsi_port_ab_io_bit(fsi, is_play);
+ ctrl = is_play ? DOFF_CTL : DIFF_CTL;
+
+ /* set IMSK */
+ fsi_irq_disable(fsi, is_play);
+
+ /* set interrupt generation factor */
+ fsi_reg_write(fsi, ctrl, IRQ_HALF);
+
+ /* clear FIFO */
+ fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR);
+
+ /* clear interrupt factor */
+ fsi_master_mask_set(INT_ST, data, 0);
+}
+
+static void fsi_soft_all_reset(void)
+{
+ u32 status = fsi_master_read(SOFT_RST);
+
+ /* port AB reset */
+ status &= 0x000000ff;
+ fsi_master_write(SOFT_RST, status);
+ mdelay(10);
+
+ /* soft reset */
+ status &= 0x000000f0;
+ fsi_master_write(SOFT_RST, status);
+ status |= 0x00000001;
+ fsi_master_write(SOFT_RST, status);
+ mdelay(10);
+}
+
+static void fsi_16data_push(struct fsi_priv *fsi,
+ struct snd_pcm_runtime *runtime,
+ int send)
+{
+ u16 *dma_start;
+ u32 snd;
+ int i;
+
+ /* get dma start position for FSI */
+ dma_start = (u16 *)runtime->dma_area;
+ dma_start += fsi->byte_offset / 2;
+
+ /*
+ * soft dma
+ * FSI can not use DMA when 16bpp
+ */
+ for (i = 0; i < send; i++) {
+ snd = (u32)dma_start[i];
+ fsi_reg_write(fsi, DODT, snd << 8);
+ }
+}
+
+static void fsi_32data_push(struct fsi_priv *fsi,
+ struct snd_pcm_runtime *runtime,
+ int send)
+{
+ u32 *dma_start;
+
+ /* get dma start position for FSI */
+ dma_start = (u32 *)runtime->dma_area;
+ dma_start += fsi->byte_offset / 4;
+
+ dma_wait_for_completion(fsi->dma_chan);
+ dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR));
+ dma_write(fsi->dma_chan, (u32)dma_start,
+ (u32)(fsi->base + DODT), send * 4);
+}
+
+/* playback interrupt */
+static int fsi_data_push(struct fsi_priv *fsi)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_substream *substream = NULL;
+ int send;
+ int fifo_free;
+ int width;
+
+ if (!fsi ||
+ !fsi->substream ||
+ !fsi->substream->runtime)
+ return -EINVAL;
+
+ runtime = fsi->substream->runtime;
+
+ /* FSI FIFO has limit.
+ * So, this driver can not send periods data at a time
+ */
+ if (fsi->byte_offset >=
+ fsi->period_len * (fsi->periods + 1)) {
+
+ substream = fsi->substream;
+ fsi->periods = (fsi->periods + 1) % runtime->periods;
+
+ if (0 == fsi->periods)
+ fsi->byte_offset = 0;
+ }
+
+ /* get 1 channel data width */
+ width = frames_to_bytes(runtime, 1) / fsi->chan;
+
+ /* get send size for alsa */
+ send = (fsi->buffer_len - fsi->byte_offset) / width;
+
+ /* get FIFO free size */
+ fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1);
+
+ /* size check */
+ if (fifo_free < send)
+ send = fifo_free;
+
+ if (2 == width)
+ fsi_16data_push(fsi, runtime, send);
+ else if (4 == width)
+ fsi_32data_push(fsi, runtime, send);
+ else
+ return -EINVAL;
+
+ fsi->byte_offset += send * width;
+
+ fsi_irq_enable(fsi, 1);
+
+ if (substream)
+ snd_pcm_period_elapsed(substream);
+
+ return 0;
+}
+
+static irqreturn_t fsi_interrupt(int irq, void *data)
+{
+ u32 status = fsi_master_read(SOFT_RST) & ~0x00000010;
+ u32 int_st = fsi_master_read(INT_ST);
+
+ /* clear irq status */
+ fsi_master_write(SOFT_RST, status);
+ fsi_master_write(SOFT_RST, status | 0x00000010);
+
+ if (int_st & INT_A_OUT)
+ fsi_data_push(&master->fsia);
+ if (int_st & INT_B_OUT)
+ fsi_data_push(&master->fsib);
+
+ fsi_master_write(INT_ST, 0x0000000);
+
+ return IRQ_HANDLED;
+}
+
+/************************************************************************
+
+
+ dai ops
+
+
+************************************************************************/
+static int fsi_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ const char *msg;
+ u32 flags = fsi_get_info_flags(fsi);
+ u32 fmt;
+ u32 reg;
+ u32 data;
+ int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ int is_master;
+ int ret = 0;
+
+ clk_enable(master->clk);
+
+ /* CKG1 */
+ data = is_play ? (1 << 0) : (1 << 4);
+ is_master = fsi_is_master_mode(fsi, is_play);
+ if (is_master)
+ fsi_reg_mask_set(fsi, CKG1, data, data);
+ else
+ fsi_reg_mask_set(fsi, CKG1, data, 0);
+
+ /* clock inversion (CKG2) */
+ data = 0;
+ switch (SH_FSI_INVERSION_MASK & flags) {
+ case SH_FSI_LRM_INV:
+ data = 1 << 12;
+ break;
+ case SH_FSI_BRM_INV:
+ data = 1 << 8;
+ break;
+ case SH_FSI_LRS_INV:
+ data = 1 << 4;
+ break;
+ case SH_FSI_BRS_INV:
+ data = 1 << 0;
+ break;
+ }
+ fsi_reg_write(fsi, CKG2, data);
+
+ /* do fmt, di fmt */
+ data = 0;
+ reg = is_play ? DO_FMT : DI_FMT;
+ fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
+ switch (fmt) {
+ case SH_FSI_FMT_MONO:
+ msg = "MONO";
+ data = CR_FMT(CR_MONO);
+ fsi->chan = 1;
+ break;
+ case SH_FSI_FMT_MONO_DELAY:
+ msg = "MONO Delay";
+ data = CR_FMT(CR_MONO_D);
+ fsi->chan = 1;
+ break;
+ case SH_FSI_FMT_PCM:
+ msg = "PCM";
+ data = CR_FMT(CR_PCM);
+ fsi->chan = 2;
+ break;
+ case SH_FSI_FMT_I2S:
+ msg = "I2S";
+ data = CR_FMT(CR_I2S);
+ fsi->chan = 2;
+ break;
+ case SH_FSI_FMT_TDM:
+ msg = "TDM";
+ data = CR_FMT(CR_TDM) | (fsi->chan - 1);
+ fsi->chan = is_play ?
+ SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+ break;
+ case SH_FSI_FMT_TDM_DELAY:
+ msg = "TDM Delay";
+ data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
+ fsi->chan = is_play ?
+ SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+ break;
+ default:
+ dev_err(dai->dev, "unknown format.\n");
+ return -EINVAL;
+ }
+
+ switch (fsi->chan) {
+ case 1:
+ fsi->fifo_max = 256;
+ break;
+ case 2:
+ fsi->fifo_max = 128;
+ break;
+ case 3:
+ case 4:
+ fsi->fifo_max = 64;
+ break;
+ case 5:
+ case 6:
+ case 7:
+ case 8:
+ fsi->fifo_max = 32;
+ break;
+ default:
+ dev_err(dai->dev, "channel size error.\n");
+ return -EINVAL;
+ }
+
+ fsi_reg_write(fsi, reg, data);
+ dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n",
+ msg, fsi->chan, fsi->dma_chan);
+
+ /*
+ * clear clk reset if master mode
+ */
+ if (is_master)
+ fsi_clk_ctrl(fsi, 1);
+
+ /* irq setting */
+ fsi_irq_init(fsi, is_play);
+
+ return ret;
+}
+
+static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ fsi_irq_disable(fsi, is_play);
+ fsi_clk_ctrl(fsi, 0);
+
+ clk_disable(master->clk);
+}
+
+static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct fsi_priv *fsi = fsi_get(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int ret = 0;
+
+ /* capture not supported */
+ if (!is_play)
+ return -ENODEV;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ fsi_stream_push(fsi, substream,
+ frames_to_bytes(runtime, runtime->buffer_size),
+ frames_to_bytes(runtime, runtime->period_size));
+ ret = fsi_data_push(fsi);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ fsi_irq_disable(fsi, is_play);
+ fsi_stream_pop(fsi);
+ break;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_ops fsi_dai_ops = {
+ .startup = fsi_dai_startup,
+ .shutdown = fsi_dai_shutdown,
+ .trigger = fsi_dai_trigger,
+};
+
+/************************************************************************
+
+
+ pcm ops
+
+
+************************************************************************/
+static struct snd_pcm_hardware fsi_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = FSI_FMTS,
+ .rates = FSI_RATES,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 1,
+ .periods_max = 32,
+ .fifo_size = 256,
+};
+
+static int fsi_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+
+ return ret;
+}
+
+static int fsi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int fsi_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsi_priv *fsi = fsi_get(substream);
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ long location;
+
+ location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play);
+ if (location < 0)
+ location = 0;
+
+ return bytes_to_frames(runtime, location);
+}
+
+static struct snd_pcm_ops fsi_pcm_ops = {
+ .open = fsi_pcm_open,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsi_hw_params,
+ .hw_free = fsi_hw_free,
+ .pointer = fsi_pointer,
+};
+
+/************************************************************************
+
+
+ snd_soc_platform
+
+
+************************************************************************/
+#define PREALLOC_BUFFER (32 * 1024)
+#define PREALLOC_BUFFER_MAX (32 * 1024)
+
+static void fsi_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int fsi_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ /*
+ * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
+ * in MMAP mode (i.e. aplay -M)
+ */
+ return snd_pcm_lib_preallocate_pages_for_all(
+ pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ PREALLOC_BUFFER, PREALLOC_BUFFER_MAX);
+}
+
+/************************************************************************
+
+
+ alsa struct
+
+
+************************************************************************/
+struct snd_soc_dai fsi_soc_dai[] = {
+ {
+ .name = "FSIA",
+ .id = 0,
+ .playback = {
+ .rates = FSI_RATES,
+ .formats = FSI_FMTS,
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ /* capture not supported */
+ .ops = &fsi_dai_ops,
+ },
+ {
+ .name = "FSIB",
+ .id = 1,
+ .playback = {
+ .rates = FSI_RATES,
+ .formats = FSI_FMTS,
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ /* capture not supported */
+ .ops = &fsi_dai_ops,
+ },
+};
+EXPORT_SYMBOL_GPL(fsi_soc_dai);
+
+struct snd_soc_platform fsi_soc_platform = {
+ .name = "fsi-pcm",
+ .pcm_ops = &fsi_pcm_ops,
+ .pcm_new = fsi_pcm_new,
+ .pcm_free = fsi_pcm_free,
+};
+EXPORT_SYMBOL_GPL(fsi_soc_platform);
+
+/************************************************************************
+
+
+ platform function
+
+
+************************************************************************/
+static int fsi_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ char clk_name[8];
+ unsigned int irq;
+ int ret;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ irq = platform_get_irq(pdev, 0);
+ if (!res || !irq) {
+ dev_err(&pdev->dev, "Not enough FSI platform resources.\n");
+ ret = -ENODEV;
+ goto exit;
+ }
+
+ master = kzalloc(sizeof(*master), GFP_KERNEL);
+ if (!master) {
+ dev_err(&pdev->dev, "Could not allocate master\n");
+ ret = -ENOMEM;
+ goto exit;
+ }
+
+ master->base = ioremap_nocache(res->start, resource_size(res));
+ if (!master->base) {
+ ret = -ENXIO;
+ dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n");
+ goto exit_kfree;
+ }
+
+ master->irq = irq;
+ master->info = pdev->dev.platform_data;
+ master->fsia.base = master->base;
+ master->fsib.base = master->base + 0x40;
+
+ master->fsia.dma_chan = -1;
+ master->fsib.dma_chan = -1;
+
+ ret = fsi_get_dma_chan();
+ if (ret < 0) {
+ dev_err(&pdev->dev, "cannot get dma api\n");
+ goto exit_iounmap;
+ }
+
+ /* FSI is based on SPU mstp */
+ snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id);
+ master->clk = clk_get(NULL, clk_name);
+ if (IS_ERR(master->clk)) {
+ dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name);
+ ret = -EIO;
+ goto exit_free_dma;
+ }
+
+ fsi_soc_dai[0].dev = &pdev->dev;
+ fsi_soc_dai[1].dev = &pdev->dev;
+
+ fsi_soft_all_reset();
+
+ ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master);
+ if (ret) {
+ dev_err(&pdev->dev, "irq request err\n");
+ goto exit_free_dma;
+ }
+
+ ret = snd_soc_register_platform(&fsi_soc_platform);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "cannot snd soc register\n");
+ goto exit_free_irq;
+ }
+
+ return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+
+exit_free_irq:
+ free_irq(irq, master);
+exit_free_dma:
+ fsi_free_dma_chan();
+exit_iounmap:
+ iounmap(master->base);
+exit_kfree:
+ kfree(master);
+ master = NULL;
+exit:
+ return ret;
+}
+
+static int fsi_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+ snd_soc_unregister_platform(&fsi_soc_platform);
+
+ clk_put(master->clk);
+
+ fsi_free_dma_chan();
+
+ free_irq(master->irq, master);
+
+ iounmap(master->base);
+ kfree(master);
+ master = NULL;
+ return 0;
+}
+
+static struct platform_driver fsi_driver = {
+ .driver = {
+ .name = "sh_fsi",
+ },
+ .probe = fsi_probe,
+ .remove = fsi_remove,
+};
+
+static int __init fsi_mobile_init(void)
+{
+ return platform_driver_register(&fsi_driver);
+}
+
+static void __exit fsi_mobile_exit(void)
+{
+ platform_driver_unregister(&fsi_driver);
+}
+module_init(fsi_mobile_init);
+module_exit(fsi_mobile_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
+MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
new file mode 100644
index 0000000..c8ceddc
--- /dev/null
+++ b/sound/soc/soc-cache.c
@@ -0,0 +1,218 @@
+/*
+ * soc-cache.c -- ASoC register cache helpers
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= codec->reg_cache_size)
+ return -1;
+ return cache[reg];
+}
+
+static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ u8 data[2];
+ int ret;
+
+ BUG_ON(codec->volatile_register);
+
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ if (reg < codec->reg_cache_size)
+ cache[reg] = value;
+ ret = codec->hw_write(codec->control_data, data, 2);
+ if (ret == 2)
+ return 0;
+ if (ret < 0)
+ return ret;
+ else
+ return -EIO;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int snd_soc_7_9_spi_write(void *control_data, const char *data,
+ int len)
+{
+ struct spi_device *spi = control_data;
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#else
+#define snd_soc_7_9_spi_write NULL
+#endif
+
+static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *reg_cache = codec->reg_cache;
+ u8 data[3];
+
+ data[0] = reg;
+ data[1] = (value >> 8) & 0xff;
+ data[2] = value & 0xff;
+
+ if (!snd_soc_codec_volatile_register(codec, reg))
+ reg_cache[reg] = value;
+
+ if (codec->hw_write(codec->control_data, data, 3) == 3)
+ return 0;
+ else
+ return -EIO;
+}
+
+static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= codec->reg_cache_size ||
+ snd_soc_codec_volatile_register(codec, reg))
+ return codec->hw_read(codec, reg);
+ else
+ return cache[reg];
+}
+
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ struct i2c_msg xfer[2];
+ u8 reg = r;
+ u16 data;
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = &reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret != 2) {
+ dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+#else
+#define snd_soc_8_16_read_i2c NULL
+#endif
+
+static struct {
+ int addr_bits;
+ int data_bits;
+ int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
+ int (*spi_write)(void *, const char *, int);
+ unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+} io_types[] = {
+ { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read },
+ { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read,
+ snd_soc_8_16_read_i2c },
+};
+
+/**
+ * snd_soc_codec_set_cache_io: Set up standard I/O functions.
+ *
+ * @codec: CODEC to configure.
+ * @type: Type of cache.
+ * @addr_bits: Number of bits of register address data.
+ * @data_bits: Number of bits of data per register.
+ * @control: Control bus used.
+ *
+ * Register formats are frequently shared between many I2C and SPI
+ * devices. In order to promote code reuse the ASoC core provides
+ * some standard implementations of CODEC read and write operations
+ * which can be set up using this function.
+ *
+ * The caller is responsible for allocating and initialising the
+ * actual cache.
+ *
+ * Note that at present this code cannot be used by CODECs with
+ * volatile registers.
+ */
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(io_types); i++)
+ if (io_types[i].addr_bits == addr_bits &&
+ io_types[i].data_bits == data_bits)
+ break;
+ if (i == ARRAY_SIZE(io_types)) {
+ printk(KERN_ERR
+ "No I/O functions for %d bit address %d bit data\n",
+ addr_bits, data_bits);
+ return -EINVAL;
+ }
+
+ codec->write = io_types[i].write;
+ codec->read = io_types[i].read;
+
+ switch (control) {
+ case SND_SOC_CUSTOM:
+ break;
+
+ case SND_SOC_I2C:
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+ codec->hw_write = (hw_write_t)i2c_master_send;
+#endif
+ if (io_types[i].i2c_read)
+ codec->hw_read = io_types[i].i2c_read;
+ break;
+
+ case SND_SOC_SPI:
+ if (io_types[i].spi_write)
+ codec->hw_write = io_types[i].spi_write;
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1d70829..7ff04ad 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -28,6 +28,7 @@
#include <linux/bitops.h>
#include <linux/debugfs.h>
#include <linux/platform_device.h>
+#include <sound/ac97_codec.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -619,8 +620,9 @@ static struct snd_pcm_ops soc_pcm_ops = {
#ifdef CONFIG_PM
/* powers down audio subsystem for suspend */
-static int soc_suspend(struct platform_device *pdev, pm_message_t state)
+static int soc_suspend(struct device *dev)
{
+ struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
struct snd_soc_platform *platform = card->platform;
@@ -656,7 +658,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
snd_pcm_suspend_all(card->dai_link[i].pcm);
if (card->suspend_pre)
- card->suspend_pre(pdev, state);
+ card->suspend_pre(pdev, PMSG_SUSPEND);
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
@@ -682,7 +684,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
}
if (codec_dev->suspend)
- codec_dev->suspend(pdev, state);
+ codec_dev->suspend(pdev, PMSG_SUSPEND);
for (i = 0; i < card->num_links; i++) {
struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
@@ -691,7 +693,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
}
if (card->suspend_post)
- card->suspend_post(pdev, state);
+ card->suspend_post(pdev, PMSG_SUSPEND);
return 0;
}
@@ -765,8 +767,9 @@ static void soc_resume_deferred(struct work_struct *work)
}
/* powers up audio subsystem after a suspend */
-static int soc_resume(struct platform_device *pdev)
+static int soc_resume(struct device *dev)
{
+ struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_card *card = socdev->card;
struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
@@ -788,6 +791,44 @@ static int soc_resume(struct platform_device *pdev)
return 0;
}
+/**
+ * snd_soc_suspend_device: Notify core of device suspend
+ *
+ * @dev: Device being suspended.
+ *
+ * In order to ensure that the entire audio subsystem is suspended in a
+ * coordinated fashion ASoC devices should suspend themselves when
+ * called by ASoC. When the standard kernel suspend process asks the
+ * device to suspend it should call this function to initiate a suspend
+ * of the entire ASoC card.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_suspend_device(struct device *dev)
+{
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_suspend_device);
+
+/**
+ * snd_soc_resume_device: Notify core of device resume
+ *
+ * @dev: Device being resumed.
+ *
+ * In order to ensure that the entire audio subsystem is resumed in a
+ * coordinated fashion ASoC devices should resume themselves when called
+ * by ASoC. When the standard kernel resume process asks the device
+ * to resume it should call this function. Once all the components of
+ * the card have notified that they are ready to be resumed the card
+ * will be resumed.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_resume_device(struct device *dev)
+{
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_resume_device);
#else
#define soc_suspend NULL
#define soc_resume NULL
@@ -981,16 +1022,39 @@ static int soc_remove(struct platform_device *pdev)
return 0;
}
+static int soc_poweroff(struct device *dev)
+{
+ struct platform_device *pdev = to_platform_device(dev);
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = socdev->card;
+
+ if (!card->instantiated)
+ return 0;
+
+ /* Flush out pmdown_time work - we actually do want to run it
+ * now, we're shutting down so no imminent restart. */
+ run_delayed_work(&card->delayed_work);
+
+ snd_soc_dapm_shutdown(socdev);
+
+ return 0;
+}
+
+static struct dev_pm_ops soc_pm_ops = {
+ .suspend = soc_suspend,
+ .resume = soc_resume,
+ .poweroff = soc_poweroff,
+};
+
/* ASoC platform driver */
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
.owner = THIS_MODULE,
+ .pm = &soc_pm_ops,
},
.probe = soc_probe,
.remove = soc_remove,
- .suspend = soc_suspend,
- .resume = soc_resume,
};
/* create a new pcm */
@@ -1062,6 +1126,23 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
return ret;
}
+/**
+ * snd_soc_codec_volatile_register: Report if a register is volatile.
+ *
+ * @codec: CODEC to query.
+ * @reg: Register to query.
+ *
+ * Boolean function indiciating if a CODEC register is volatile.
+ */
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
+{
+ if (codec->volatile_register)
+ return codec->volatile_register(reg);
+ else
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register);
+
/* codec register dump */
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
{
@@ -1075,6 +1156,9 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
count += sprintf(buf, "%s registers\n", codec->name);
for (i = 0; i < codec->reg_cache_size; i += step) {
+ if (codec->readable_register && !codec->readable_register(i))
+ continue;
+
count += sprintf(buf + count, "%2x: ", i);
if (count >= PAGE_SIZE - 1)
break;
@@ -1183,10 +1267,18 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
if (!codec->debugfs_pop_time)
printk(KERN_WARNING
"Failed to create pop time debugfs file\n");
+
+ codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root);
+ if (!codec->debugfs_dapm)
+ printk(KERN_WARNING
+ "Failed to create DAPM debugfs directory\n");
+
+ snd_soc_dapm_debugfs_init(codec);
}
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
{
+ debugfs_remove_recursive(codec->debugfs_dapm);
debugfs_remove(codec->debugfs_pop_time);
debugfs_remove(codec->debugfs_reg);
}
@@ -1264,10 +1356,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
* Returns 1 for change else 0.
*/
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value)
+ unsigned int mask, unsigned int value)
{
int change;
- unsigned short old, new;
+ unsigned int old, new;
mutex_lock(&io_mutex);
old = snd_soc_read(codec, reg);
@@ -1294,10 +1386,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits);
* Returns 1 for change else 0.
*/
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value)
+ unsigned int mask, unsigned int value)
{
int change;
- unsigned short old, new;
+ unsigned int old, new;
mutex_lock(&io_mutex);
old = snd_soc_read(codec, reg);
@@ -1381,8 +1473,11 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
continue;
}
}
- if (card->dai_link[i].codec_dai->ac97_control)
+ if (card->dai_link[i].codec_dai->ac97_control) {
ac97 = 1;
+ snd_ac97_dev_add_pdata(codec->ac97,
+ card->dai_link[i].cpu_dai->ac97_pdata);
+ }
}
snprintf(codec->card->shortname, sizeof(codec->card->shortname),
"%s", card->name);
@@ -1586,7 +1681,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val, bitmask;
+ unsigned int val, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
;
@@ -1615,8 +1710,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val;
- unsigned short mask, bitmask;
+ unsigned int val;
+ unsigned int mask, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
;
@@ -1652,7 +1747,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short reg_val, val, mux;
+ unsigned int reg_val, val, mux;
reg_val = snd_soc_read(codec, e->reg);
val = (reg_val >> e->shift_l) & e->mask;
@@ -1691,8 +1786,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val;
- unsigned short mask;
+ unsigned int val;
+ unsigned int mask;
if (ucontrol->value.enumerated.item[0] > e->max - 1)
return -EINVAL;
@@ -1852,7 +1947,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- unsigned short val, val2, val_mask;
+ unsigned int val, val2, val_mask;
val = (ucontrol->value.integer.value[0] & mask);
if (invert)
@@ -1918,7 +2013,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
- unsigned int mask = (1<<fls(max))-1;
+ unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
ucontrol->value.integer.value[0] =
@@ -1958,7 +2053,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
int err;
- unsigned short val, val2, val_mask;
+ unsigned int val, val2, val_mask;
val_mask = mask << shift;
val = (ucontrol->value.integer.value[0] & mask);
@@ -2050,7 +2145,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
int min = mc->min;
- unsigned short val;
+ unsigned int val;
val = (ucontrol->value.integer.value[0]+min) & 0xff;
val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
@@ -2136,17 +2231,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
/**
* snd_soc_dai_set_tdm_slot - configure DAI TDM.
* @dai: DAI
- * @mask: DAI specific mask representing used slots.
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
* @slots: Number of slots in use.
+ * @slot_width: Width in bits for each slot.
*
* Configures a DAI for TDM operation. Both mask and slots are codec and DAI
* specific.
*/
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
- unsigned int mask, int slots)
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
if (dai->ops && dai->ops->set_tdm_slot)
- return dai->ops->set_tdm_slot(dai, mask, slots);
+ return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask,
+ slots, slot_width);
else
return -EINVAL;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 21c6907..0d8b08e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -37,6 +37,7 @@
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <linux/jiffies.h>
+#include <linux/debugfs.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -52,19 +53,41 @@
/* dapm power sequences - make this per codec in the future */
static int dapm_up_seq[] = {
- snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias,
- snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
- snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl,
- snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
- snd_soc_dapm_post
+ [snd_soc_dapm_pre] = 0,
+ [snd_soc_dapm_supply] = 1,
+ [snd_soc_dapm_micbias] = 2,
+ [snd_soc_dapm_aif_in] = 3,
+ [snd_soc_dapm_aif_out] = 3,
+ [snd_soc_dapm_mic] = 4,
+ [snd_soc_dapm_mux] = 5,
+ [snd_soc_dapm_value_mux] = 5,
+ [snd_soc_dapm_dac] = 6,
+ [snd_soc_dapm_mixer] = 7,
+ [snd_soc_dapm_mixer_named_ctl] = 7,
+ [snd_soc_dapm_pga] = 8,
+ [snd_soc_dapm_adc] = 9,
+ [snd_soc_dapm_hp] = 10,
+ [snd_soc_dapm_spk] = 10,
+ [snd_soc_dapm_post] = 11,
};
static int dapm_down_seq[] = {
- snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
- snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
- snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
- snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply,
- snd_soc_dapm_post
+ [snd_soc_dapm_pre] = 0,
+ [snd_soc_dapm_adc] = 1,
+ [snd_soc_dapm_hp] = 2,
+ [snd_soc_dapm_spk] = 2,
+ [snd_soc_dapm_pga] = 4,
+ [snd_soc_dapm_mixer_named_ctl] = 5,
+ [snd_soc_dapm_mixer] = 5,
+ [snd_soc_dapm_dac] = 6,
+ [snd_soc_dapm_mic] = 7,
+ [snd_soc_dapm_micbias] = 8,
+ [snd_soc_dapm_mux] = 9,
+ [snd_soc_dapm_value_mux] = 9,
+ [snd_soc_dapm_aif_in] = 10,
+ [snd_soc_dapm_aif_out] = 10,
+ [snd_soc_dapm_supply] = 11,
+ [snd_soc_dapm_post] = 12,
};
static void pop_wait(u32 pop_time)
@@ -130,8 +153,12 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
if (card->set_bias_level)
ret = card->set_bias_level(card, level);
- if (ret == 0 && codec->set_bias_level)
- ret = codec->set_bias_level(codec, level);
+ if (ret == 0) {
+ if (codec->set_bias_level)
+ ret = codec->set_bias_level(codec, level);
+ else
+ codec->bias_level = level;
+ }
return ret;
}
@@ -206,6 +233,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
p->connect = 1;
break;
/* does effect routing - dynamically connected */
@@ -268,7 +297,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec,
static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
{
int change, power;
- unsigned short old, new;
+ unsigned int old, new;
struct snd_soc_codec *codec = widget->codec;
/* check for valid widgets */
@@ -479,8 +508,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
if (widget->id == snd_soc_dapm_supply)
return 0;
- if (widget->id == snd_soc_dapm_adc && widget->active)
- return 1;
+ switch (widget->id) {
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_aif_out:
+ if (widget->active)
+ return 1;
+ default:
+ break;
+ }
if (widget->connected) {
/* connected pin ? */
@@ -519,8 +554,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
return 0;
/* active stream ? */
- if (widget->id == snd_soc_dapm_dac && widget->active)
- return 1;
+ switch (widget->id) {
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_aif_in:
+ if (widget->active)
+ return 1;
+ default:
+ break;
+ }
if (widget->connected) {
/* connected pin ? */
@@ -689,53 +730,211 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
return power;
}
-/*
- * Scan a single DAPM widget for a complete audio path and update the
- * power status appropriately.
- */
-static int dapm_power_widget(struct snd_soc_codec *codec, int event,
- struct snd_soc_dapm_widget *w)
+static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
+ struct snd_soc_dapm_widget *b,
+ int sort[])
{
- int ret;
+ if (sort[a->id] != sort[b->id])
+ return sort[a->id] - sort[b->id];
+ if (a->reg != b->reg)
+ return a->reg - b->reg;
- switch (w->id) {
- case snd_soc_dapm_pre:
- if (!w->event)
- return 0;
+ return 0;
+}
- if (event == SND_SOC_DAPM_STREAM_START) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMU);
+/* Insert a widget in order into a DAPM power sequence. */
+static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget,
+ struct list_head *list,
+ int sort[])
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, list, power_list)
+ if (dapm_seq_compare(new_widget, w, sort) < 0) {
+ list_add_tail(&new_widget->power_list, &w->power_list);
+ return;
+ }
+
+ list_add_tail(&new_widget->power_list, list);
+}
+
+/* Apply the coalesced changes from a DAPM sequence */
+static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
+ struct list_head *pending)
+{
+ struct snd_soc_dapm_widget *w;
+ int reg, power, ret;
+ unsigned int value = 0;
+ unsigned int mask = 0;
+ unsigned int cur_mask;
+
+ reg = list_first_entry(pending, struct snd_soc_dapm_widget,
+ power_list)->reg;
+
+ list_for_each_entry(w, pending, power_list) {
+ cur_mask = 1 << w->shift;
+ BUG_ON(reg != w->reg);
+
+ if (w->invert)
+ power = !w->power;
+ else
+ power = w->power;
+
+ mask |= cur_mask;
+ if (power)
+ value |= cur_mask;
+
+ pop_dbg(codec->pop_time,
+ "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n",
+ w->name, reg, value, mask);
+
+ /* power up pre event */
+ if (w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+ pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n",
+ w->name);
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
if (ret < 0)
- return ret;
- } else if (event == SND_SOC_DAPM_STREAM_STOP) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_PRE_PMD);
+ pr_err("%s: pre event failed: %d\n",
+ w->name, ret);
+ }
+
+ /* power down pre event */
+ if (!w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+ pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n",
+ w->name);
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
if (ret < 0)
- return ret;
+ pr_err("%s: pre event failed: %d\n",
+ w->name, ret);
}
- return 0;
- case snd_soc_dapm_post:
- if (!w->event)
- return 0;
+ /* Lower PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && !w->power)
+ dapm_set_pga(w, w->power);
+ }
- if (event == SND_SOC_DAPM_STREAM_START) {
+ if (reg >= 0) {
+ pop_dbg(codec->pop_time,
+ "pop test : Applying 0x%x/0x%x to %x in %dms\n",
+ value, mask, reg, codec->pop_time);
+ pop_wait(codec->pop_time);
+ snd_soc_update_bits(codec, reg, mask, value);
+ }
+
+ list_for_each_entry(w, pending, power_list) {
+ /* Raise PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && w->power)
+ dapm_set_pga(w, w->power);
+
+ /* power up post event */
+ if (w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+ pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n",
+ w->name);
ret = w->event(w,
NULL, SND_SOC_DAPM_POST_PMU);
if (ret < 0)
- return ret;
- } else if (event == SND_SOC_DAPM_STREAM_STOP) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMD);
+ pr_err("%s: post event failed: %d\n",
+ w->name, ret);
+ }
+
+ /* power down post event */
+ if (!w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+ pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n",
+ w->name);
+ ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
if (ret < 0)
- return ret;
+ pr_err("%s: post event failed: %d\n",
+ w->name, ret);
}
- return 0;
+ }
+}
- default:
- return dapm_generic_apply_power(w);
+/* Apply a DAPM power sequence.
+ *
+ * We walk over a pre-sorted list of widgets to apply power to. In
+ * order to minimise the number of writes to the device required
+ * multiple widgets will be updated in a single write where possible.
+ * Currently anything that requires more than a single write is not
+ * handled.
+ */
+static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
+ int event, int sort[])
+{
+ struct snd_soc_dapm_widget *w, *n;
+ LIST_HEAD(pending);
+ int cur_sort = -1;
+ int cur_reg = SND_SOC_NOPM;
+ int ret;
+
+ list_for_each_entry_safe(w, n, list, power_list) {
+ ret = 0;
+
+ /* Do we need to apply any queued changes? */
+ if (sort[w->id] != cur_sort || w->reg != cur_reg) {
+ if (!list_empty(&pending))
+ dapm_seq_run_coalesced(codec, &pending);
+
+ INIT_LIST_HEAD(&pending);
+ cur_sort = -1;
+ cur_reg = SND_SOC_NOPM;
+ }
+
+ switch (w->id) {
+ case snd_soc_dapm_pre:
+ if (!w->event)
+ list_for_each_entry_safe_continue(w, n, list,
+ power_list);
+
+ if (event == SND_SOC_DAPM_STREAM_START)
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMU);
+ else if (event == SND_SOC_DAPM_STREAM_STOP)
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMD);
+ break;
+
+ case snd_soc_dapm_post:
+ if (!w->event)
+ list_for_each_entry_safe_continue(w, n, list,
+ power_list);
+
+ if (event == SND_SOC_DAPM_STREAM_START)
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ else if (event == SND_SOC_DAPM_STREAM_STOP)
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMD);
+ break;
+
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_spk:
+ /* No register support currently */
+ ret = dapm_generic_apply_power(w);
+ break;
+
+ default:
+ /* Queue it up for application */
+ cur_sort = sort[w->id];
+ cur_reg = w->reg;
+ list_move(&w->power_list, &pending);
+ break;
+ }
+
+ if (ret < 0)
+ pr_err("Failed to apply widget power: %d\n",
+ ret);
}
+
+ if (!list_empty(&pending))
+ dapm_seq_run_coalesced(codec, &pending);
}
/*
@@ -751,23 +950,22 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
{
struct snd_soc_device *socdev = codec->socdev;
struct snd_soc_dapm_widget *w;
+ LIST_HEAD(up_list);
+ LIST_HEAD(down_list);
int ret = 0;
- int i, power;
+ int power;
int sys_power = 0;
- INIT_LIST_HEAD(&codec->up_list);
- INIT_LIST_HEAD(&codec->down_list);
-
/* Check which widgets we need to power and store them in
* lists indicating if they should be powered up or down.
*/
list_for_each_entry(w, &codec->dapm_widgets, list) {
switch (w->id) {
case snd_soc_dapm_pre:
- list_add_tail(&codec->down_list, &w->power_list);
+ dapm_seq_insert(w, &down_list, dapm_down_seq);
break;
case snd_soc_dapm_post:
- list_add_tail(&codec->up_list, &w->power_list);
+ dapm_seq_insert(w, &up_list, dapm_up_seq);
break;
default:
@@ -782,16 +980,31 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
continue;
if (power)
- list_add_tail(&w->power_list, &codec->up_list);
+ dapm_seq_insert(w, &up_list, dapm_up_seq);
else
- list_add_tail(&w->power_list,
- &codec->down_list);
+ dapm_seq_insert(w, &down_list, dapm_down_seq);
w->power = power;
break;
}
}
+ /* If there are no DAPM widgets then try to figure out power from the
+ * event type.
+ */
+ if (list_empty(&codec->dapm_widgets)) {
+ switch (event) {
+ case SND_SOC_DAPM_STREAM_START:
+ case SND_SOC_DAPM_STREAM_RESUME:
+ sys_power = 1;
+ break;
+ case SND_SOC_DAPM_STREAM_NOP:
+ sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+ default:
+ break;
+ }
+ }
+
/* If we're changing to all on or all off then prepare */
if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) ||
(!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) {
@@ -802,32 +1015,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
}
/* Power down widgets first; try to avoid amplifying pops. */
- for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) {
- list_for_each_entry(w, &codec->down_list, power_list) {
- /* is widget in stream order */
- if (w->id != dapm_down_seq[i])
- continue;
-
- ret = dapm_power_widget(codec, event, w);
- if (ret != 0)
- pr_err("Failed to power down %s: %d\n",
- w->name, ret);
- }
- }
+ dapm_seq_run(codec, &down_list, event, dapm_down_seq);
/* Now power up. */
- for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) {
- list_for_each_entry(w, &codec->up_list, power_list) {
- /* is widget in stream order */
- if (w->id != dapm_up_seq[i])
- continue;
-
- ret = dapm_power_widget(codec, event, w);
- if (ret != 0)
- pr_err("Failed to power up %s: %d\n",
- w->name, ret);
- }
- }
+ dapm_seq_run(codec, &up_list, event, dapm_up_seq);
/* If we just powered the last thing off drop to standby bias */
if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
@@ -845,6 +1036,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
pr_err("Failed to apply active bias: %d\n", ret);
}
+ pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n",
+ codec->pop_time);
+
return 0;
}
@@ -881,6 +1075,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
if (w->name) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
@@ -906,6 +1102,92 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
}
#endif
+#ifdef CONFIG_DEBUG_FS
+static int dapm_widget_power_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t dapm_widget_power_read_file(struct file *file,
+ char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ struct snd_soc_dapm_widget *w = file->private_data;
+ char *buf;
+ int in, out;
+ ssize_t ret;
+ struct snd_soc_dapm_path *p = NULL;
+
+ buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+
+ ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n",
+ w->name, w->power ? "On" : "Off", in, out);
+
+ if (w->active && w->sname)
+ ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
+ w->sname);
+
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connect)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ " in %s %s\n",
+ p->name ? p->name : "static",
+ p->source->name);
+ }
+ list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connect)
+ ret += snprintf(buf + ret, PAGE_SIZE - ret,
+ " out %s %s\n",
+ p->name ? p->name : "static",
+ p->sink->name);
+ }
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+ kfree(buf);
+ return ret;
+}
+
+static const struct file_operations dapm_widget_power_fops = {
+ .open = dapm_widget_power_open_file,
+ .read = dapm_widget_power_read_file,
+};
+
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w;
+ struct dentry *d;
+
+ if (!codec->debugfs_dapm)
+ return;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!w->name)
+ continue;
+
+ d = debugfs_create_file(w->name, 0444,
+ codec->debugfs_dapm, w,
+ &dapm_widget_power_fops);
+ if (!d)
+ printk(KERN_WARNING
+ "ASoC: Failed to create %s debugfs file\n",
+ w->name);
+ }
+}
+#else
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+{
+}
+#endif
+
/* test and update the power status of a mux widget */
static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int mask,
@@ -1138,8 +1420,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
if (wsink->id == snd_soc_dapm_input) {
if (wsource->id == snd_soc_dapm_micbias ||
wsource->id == snd_soc_dapm_mic ||
- wsink->id == snd_soc_dapm_line ||
- wsink->id == snd_soc_dapm_output)
+ wsource->id == snd_soc_dapm_line ||
+ wsource->id == snd_soc_dapm_output)
wsink->ext = 1;
}
if (wsource->id == snd_soc_dapm_output) {
@@ -1171,6 +1453,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
case snd_soc_dapm_supply:
+ case snd_soc_dapm_aif_in:
+ case snd_soc_dapm_aif_out:
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
@@ -1273,9 +1557,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
dapm_new_mux(codec, w);
break;
case snd_soc_dapm_adc:
+ case snd_soc_dapm_aif_out:
w->power_check = dapm_adc_check_power;
break;
case snd_soc_dapm_dac:
+ case snd_soc_dapm_aif_in:
w->power_check = dapm_dac_check_power;
break;
case snd_soc_dapm_pga:
@@ -1372,7 +1658,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- unsigned short val, val2, val_mask;
+ unsigned int val, val2, val_mask;
int ret;
val = (ucontrol->value.integer.value[0] & mask);
@@ -1436,7 +1722,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val, bitmask;
+ unsigned int val, bitmask;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
;
@@ -1464,8 +1750,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val, mux;
- unsigned short mask, bitmask;
+ unsigned int val, mux;
+ unsigned int mask, bitmask;
int ret = 0;
for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
@@ -1523,7 +1809,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short reg_val, val, mux;
+ unsigned int reg_val, val, mux;
reg_val = snd_soc_read(widget->codec, e->reg);
val = (reg_val >> e->shift_l) & e->mask;
@@ -1563,8 +1849,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val, mux;
- unsigned short mask;
+ unsigned int val, mux;
+ unsigned int mask;
int ret = 0;
if (ucontrol->value.enumerated.item[0] > e->max - 1)
@@ -1880,6 +2166,36 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev)
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
+/*
+ * snd_soc_dapm_shutdown - callback for system shutdown
+ */
+void snd_soc_dapm_shutdown(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct snd_soc_dapm_widget *w;
+ LIST_HEAD(down_list);
+ int powerdown = 0;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (w->power) {
+ dapm_seq_insert(w, &down_list, dapm_down_seq);
+ w->power = 0;
+ powerdown = 1;
+ }
+ }
+
+ /* If there were no widgets to power down we're already in
+ * standby.
+ */
+ if (powerdown) {
+ snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE);
+ dapm_seq_run(codec, &down_list, 0, dapm_down_seq);
+ snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY);
+ }
+
+ snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF);
+}
+
/* Module information */
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 28346fb..1d455ab7 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -73,14 +73,15 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
oldstatus = jack->status;
jack->status &= ~mask;
- jack->status |= status;
+ jack->status |= status & mask;
- /* The DAPM sync is expensive enough to be worth skipping */
- if (jack->status == oldstatus)
+ /* The DAPM sync is expensive enough to be worth skipping.
+ * However, empty mask means pin synchronization is desired. */
+ if (mask && (jack->status == oldstatus))
goto out;
list_for_each_entry(pin, &jack->pins, list) {
- enable = pin->mask & status;
+ enable = pin->mask & jack->status;
if (pin->invert)
enable = !enable;
@@ -220,6 +221,9 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
if (ret)
goto err;
+ INIT_WORK(&gpios[i].work, gpio_work);
+ gpios[i].jack = jack;
+
ret = request_irq(gpio_to_irq(gpios[i].gpio),
gpio_handler,
IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
@@ -228,8 +232,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
if (ret)
goto err;
- INIT_WORK(&gpios[i].work, gpio_work);
- gpios[i].jack = jack;
+#ifdef CONFIG_GPIO_SYSFS
+ /* Expose GPIO value over sysfs for diagnostic purposes */
+ gpio_export(gpios[i].gpio, false);
+#endif
+
+ /* Update initial jack status */
+ snd_soc_jack_gpio_detect(&gpios[i]);
}
return 0;
@@ -258,6 +267,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
int i;
for (i = 0; i < count; i++) {
+#ifdef CONFIG_GPIO_SYSFS
+ gpio_unexport(gpios[i].gpio);
+#endif
free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]);
gpio_free(gpios[i].gpio);
gpios[i].jack = NULL;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 938a58a..efed64b 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
static bool filter(struct dma_chan *chan, void *param)
{
struct txx9aclc_dmadata *dmadata = param;
- char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */
+ char *devname;
+ bool found = false;
- snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name,
+ devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name,
(int)dmadata->dma_res->start);
if (strcmp(dev_name(chan->device->dev), devname) == 0) {
chan->private = &dmadata->dma_slave;
- return true;
+ found = true;
}
- return false;
+ kfree(devname);
+ return found;
}
static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
diff --git a/sound/sound_core.c b/sound/sound_core.c
index a41f8b1..bb4b88e 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -128,6 +128,46 @@ extern int msnd_pinnacle_init(void);
#endif
/*
+ * By default, OSS sound_core claims full legacy minor range (0-255)
+ * of SOUND_MAJOR to trap open attempts to any sound minor and
+ * requests modules using custom sound-slot/service-* module aliases.
+ * The only benefit of doing this is allowing use of custom module
+ * aliases instead of the standard char-major-* ones. This behavior
+ * prevents alternative OSS implementation and is scheduled to be
+ * removed.
+ *
+ * CONFIG_SOUND_OSS_CORE_PRECLAIM and soundcore.preclaim_oss kernel
+ * parameter are added to allow distros and developers to try and
+ * switch to alternative implementations without needing to rebuild
+ * the kernel in the meantime. If preclaim_oss is non-zero, the
+ * kernel will behave the same as before. All SOUND_MAJOR minors are
+ * preclaimed and the custom module aliases along with standard chrdev
+ * ones are emitted if a missing device is opened. If preclaim_oss is
+ * zero, sound_core only grabs what's actually in use and for missing
+ * devices only the standard chrdev aliases are requested.
+ *
+ * All these clutters are scheduled to be removed along with
+ * sound-slot/service-* module aliases. Please take a look at
+ * feature-removal-schedule.txt for details.
+ */
+#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM
+static int preclaim_oss = 1;
+#else
+static int preclaim_oss = 0;
+#endif
+
+module_param(preclaim_oss, int, 0444);
+
+static int soundcore_open(struct inode *, struct file *);
+
+static const struct file_operations soundcore_fops =
+{
+ /* We must have an owner or the module locking fails */
+ .owner = THIS_MODULE,
+ .open = soundcore_open,
+};
+
+/*
* Low level list operator. Scan the ordered list, find a hole and
* join into it. Called with the lock asserted
*/
@@ -219,8 +259,9 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati
if (!s)
return -ENOMEM;
-
+
spin_lock(&sound_loader_lock);
+retry:
r = __sound_insert_unit(s, list, fops, index, low, top);
spin_unlock(&sound_loader_lock);
@@ -231,11 +272,31 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati
else
sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP);
+ if (!preclaim_oss) {
+ /*
+ * Something else might have grabbed the minor. If
+ * first free slot is requested, rescan with @low set
+ * to the next unit; otherwise, -EBUSY.
+ */
+ r = __register_chrdev(SOUND_MAJOR, s->unit_minor, 1, s->name,
+ &soundcore_fops);
+ if (r < 0) {
+ spin_lock(&sound_loader_lock);
+ __sound_remove_unit(list, s->unit_minor);
+ if (index < 0) {
+ low = s->unit_minor + SOUND_STEP;
+ goto retry;
+ }
+ spin_unlock(&sound_loader_lock);
+ return -EBUSY;
+ }
+ }
+
device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor),
NULL, s->name+6);
- return r;
+ return s->unit_minor;
- fail:
+fail:
kfree(s);
return r;
}
@@ -254,6 +315,9 @@ static void sound_remove_unit(struct sound_unit **list, int unit)
p = __sound_remove_unit(list, unit);
spin_unlock(&sound_loader_lock);
if (p) {
+ if (!preclaim_oss)
+ __unregister_chrdev(SOUND_MAJOR, p->unit_minor, 1,
+ p->name);
device_destroy(sound_class, MKDEV(SOUND_MAJOR, p->unit_minor));
kfree(p);
}
@@ -491,19 +555,6 @@ void unregister_sound_dsp(int unit)
EXPORT_SYMBOL(unregister_sound_dsp);
-/*
- * Now our file operations
- */
-
-static int soundcore_open(struct inode *, struct file *);
-
-static const struct file_operations soundcore_fops=
-{
- /* We must have an owner or the module locking fails */
- .owner = THIS_MODULE,
- .open = soundcore_open,
-};
-
static struct sound_unit *__look_for_unit(int chain, int unit)
{
struct sound_unit *s;
@@ -539,8 +590,9 @@ static int soundcore_open(struct inode *inode, struct file *file)
s = __look_for_unit(chain, unit);
if (s)
new_fops = fops_get(s->unit_fops);
- if (!new_fops) {
+ if (preclaim_oss && !new_fops) {
spin_unlock(&sound_loader_lock);
+
/*
* Please, don't change this order or code.
* For ALSA slot means soundcard and OSS emulation code
@@ -550,6 +602,17 @@ static int soundcore_open(struct inode *inode, struct file *file)
*/
request_module("sound-slot-%i", unit>>4);
request_module("sound-service-%i-%i", unit>>4, chain);
+
+ /*
+ * sound-slot/service-* module aliases are scheduled
+ * for removal in favor of the standard char-major-*
+ * module aliases. For the time being, generate both
+ * the legacy and standard module aliases to ease
+ * transition.
+ */
+ if (request_module("char-major-%d-%d", SOUND_MAJOR, unit) > 0)
+ request_module("char-major-%d", SOUND_MAJOR);
+
spin_lock(&sound_loader_lock);
s = __look_for_unit(chain, unit);
if (s)
@@ -593,7 +656,8 @@ static void cleanup_oss_soundcore(void)
static int __init init_oss_soundcore(void)
{
- if (register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops)==-1) {
+ if (preclaim_oss &&
+ register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) == -1) {
printk(KERN_ERR "soundcore: sound device already in use.\n");
return -EBUSY;
}
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 44b9cdc..8db0374 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1083,6 +1083,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
} else
urb_packs = 1;
urb_packs *= packs_per_ms;
+ if (subs->syncpipe)
+ urb_packs = min(urb_packs, 1U << subs->syncinterval);
/* decide how many packets to be used */
if (is_playback) {
@@ -2124,8 +2126,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
fp = list_entry(p, struct audioformat, list);
snd_iprintf(buffer, " Interface %d\n", fp->iface);
snd_iprintf(buffer, " Altset %d\n", fp->altsetting);
- snd_iprintf(buffer, " Format: %#x (%d bits)\n",
- fp->format, snd_pcm_format_width(fp->format));
+ snd_iprintf(buffer, " Format: %s\n",
+ snd_pcm_format_name(fp->format));
snd_iprintf(buffer, " Channels: %d\n", fp->channels);
snd_iprintf(buffer, " Endpoint: %d %s (%s)\n",
fp->endpoint & USB_ENDPOINT_NUMBER_MASK,
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 2fb35cc..0eff19c 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -45,6 +45,7 @@
#include <linux/slab.h>
#include <linux/timer.h>
#include <linux/usb.h>
+#include <linux/wait.h>
#include <sound/core.h>
#include <sound/rawmidi.h>
#include <sound/asequencer.h>
@@ -62,6 +63,9 @@
*/
#define ERROR_DELAY_JIFFIES (HZ / 10)
+#define OUTPUT_URBS 7
+#define INPUT_URBS 7
+
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_DESCRIPTION("USB Audio/MIDI helper module");
@@ -90,7 +94,7 @@ struct snd_usb_midi_endpoint;
struct usb_protocol_ops {
void (*input)(struct snd_usb_midi_in_endpoint*, uint8_t*, int);
- void (*output)(struct snd_usb_midi_out_endpoint*);
+ void (*output)(struct snd_usb_midi_out_endpoint *ep, struct urb *urb);
void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t);
void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint*);
void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint*);
@@ -116,11 +120,15 @@ struct snd_usb_midi {
struct snd_usb_midi_out_endpoint {
struct snd_usb_midi* umidi;
- struct urb* urb;
- int urb_active;
+ struct out_urb_context {
+ struct urb *urb;
+ struct snd_usb_midi_out_endpoint *ep;
+ } urbs[OUTPUT_URBS];
+ unsigned int active_urbs;
+ unsigned int drain_urbs;
int max_transfer; /* size of urb buffer */
struct tasklet_struct tasklet;
-
+ unsigned int next_urb;
spinlock_t buffer_lock;
struct usbmidi_out_port {
@@ -139,11 +147,13 @@ struct snd_usb_midi_out_endpoint {
uint8_t data[2];
} ports[0x10];
int current_port;
+
+ wait_queue_head_t drain_wait;
};
struct snd_usb_midi_in_endpoint {
struct snd_usb_midi* umidi;
- struct urb* urb;
+ struct urb* urbs[INPUT_URBS];
struct usbmidi_in_port {
struct snd_rawmidi_substream *substream;
u8 running_status_length;
@@ -251,10 +261,17 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb)
static void snd_usbmidi_out_urb_complete(struct urb* urb)
{
- struct snd_usb_midi_out_endpoint* ep = urb->context;
+ struct out_urb_context *context = urb->context;
+ struct snd_usb_midi_out_endpoint* ep = context->ep;
+ unsigned int urb_index;
spin_lock(&ep->buffer_lock);
- ep->urb_active = 0;
+ urb_index = context - ep->urbs;
+ ep->active_urbs &= ~(1 << urb_index);
+ if (unlikely(ep->drain_urbs)) {
+ ep->drain_urbs &= ~(1 << urb_index);
+ wake_up(&ep->drain_wait);
+ }
spin_unlock(&ep->buffer_lock);
if (urb->status < 0) {
int err = snd_usbmidi_urb_error(urb->status);
@@ -274,24 +291,38 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb)
*/
static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep)
{
- struct urb* urb = ep->urb;
+ unsigned int urb_index;
+ struct urb* urb;
unsigned long flags;
spin_lock_irqsave(&ep->buffer_lock, flags);
- if (ep->urb_active || ep->umidi->chip->shutdown) {
+ if (ep->umidi->chip->shutdown) {
spin_unlock_irqrestore(&ep->buffer_lock, flags);
return;
}
- urb->transfer_buffer_length = 0;
- ep->umidi->usb_protocol_ops->output(ep);
+ urb_index = ep->next_urb;
+ for (;;) {
+ if (!(ep->active_urbs & (1 << urb_index))) {
+ urb = ep->urbs[urb_index].urb;
+ urb->transfer_buffer_length = 0;
+ ep->umidi->usb_protocol_ops->output(ep, urb);
+ if (urb->transfer_buffer_length == 0)
+ break;
- if (urb->transfer_buffer_length > 0) {
- dump_urb("sending", urb->transfer_buffer,
- urb->transfer_buffer_length);
- urb->dev = ep->umidi->chip->dev;
- ep->urb_active = snd_usbmidi_submit_urb(urb, GFP_ATOMIC) >= 0;
+ dump_urb("sending", urb->transfer_buffer,
+ urb->transfer_buffer_length);
+ urb->dev = ep->umidi->chip->dev;
+ if (snd_usbmidi_submit_urb(urb, GFP_ATOMIC) < 0)
+ break;
+ ep->active_urbs |= 1 << urb_index;
+ }
+ if (++urb_index >= OUTPUT_URBS)
+ urb_index = 0;
+ if (urb_index == ep->next_urb)
+ break;
}
+ ep->next_urb = urb_index;
spin_unlock_irqrestore(&ep->buffer_lock, flags);
}
@@ -306,7 +337,7 @@ static void snd_usbmidi_out_tasklet(unsigned long data)
static void snd_usbmidi_error_timer(unsigned long data)
{
struct snd_usb_midi *umidi = (struct snd_usb_midi *)data;
- int i;
+ unsigned int i, j;
spin_lock(&umidi->disc_lock);
if (umidi->disconnected) {
@@ -317,8 +348,10 @@ static void snd_usbmidi_error_timer(unsigned long data)
struct snd_usb_midi_in_endpoint *in = umidi->endpoints[i].in;
if (in && in->error_resubmit) {
in->error_resubmit = 0;
- in->urb->dev = umidi->chip->dev;
- snd_usbmidi_submit_urb(in->urb, GFP_ATOMIC);
+ for (j = 0; j < INPUT_URBS; ++j) {
+ in->urbs[j]->dev = umidi->chip->dev;
+ snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC);
+ }
}
if (umidi->endpoints[i].out)
snd_usbmidi_do_output(umidi->endpoints[i].out);
@@ -330,13 +363,14 @@ static void snd_usbmidi_error_timer(unsigned long data)
static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep,
const void *data, int len)
{
- int err;
+ int err = 0;
void *buf = kmemdup(data, len, GFP_KERNEL);
if (!buf)
return -ENOMEM;
dump_urb("sending", buf, len);
- err = usb_bulk_msg(ep->umidi->chip->dev, ep->urb->pipe, buf, len,
- NULL, 250);
+ if (ep->urbs[0].urb)
+ err = usb_bulk_msg(ep->umidi->chip->dev, ep->urbs[0].urb->pipe,
+ buf, len, NULL, 250);
kfree(buf);
return err;
}
@@ -554,9 +588,9 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port,
}
}
-static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep,
+ struct urb *urb)
{
- struct urb* urb = ep->urb;
int p;
/* FIXME: lower-numbered ports can starve higher-numbered ports */
@@ -613,14 +647,15 @@ static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint* ep,
snd_usbmidi_input_data(ep, 0, &buffer[2], buffer[0] - 1);
}
-static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep,
+ struct urb *urb)
{
uint8_t* transfer_buffer;
int count;
if (!ep->ports[0].active)
return;
- transfer_buffer = ep->urb->transfer_buffer;
+ transfer_buffer = urb->transfer_buffer;
count = snd_rawmidi_transmit(ep->ports[0].substream,
&transfer_buffer[2],
ep->max_transfer - 2);
@@ -630,7 +665,7 @@ static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep)
}
transfer_buffer[0] = 0;
transfer_buffer[1] = count;
- ep->urb->transfer_buffer_length = 2 + count;
+ urb->transfer_buffer_length = 2 + count;
}
static struct usb_protocol_ops snd_usbmidi_novation_ops = {
@@ -648,20 +683,21 @@ static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep,
snd_usbmidi_input_data(ep, 0, buffer, buffer_length);
}
-static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep,
+ struct urb *urb)
{
int count;
if (!ep->ports[0].active)
return;
count = snd_rawmidi_transmit(ep->ports[0].substream,
- ep->urb->transfer_buffer,
+ urb->transfer_buffer,
ep->max_transfer);
if (count < 1) {
ep->ports[0].active = 0;
return;
}
- ep->urb->transfer_buffer_length = count;
+ urb->transfer_buffer_length = count;
}
static struct usb_protocol_ops snd_usbmidi_raw_ops = {
@@ -681,23 +717,25 @@ static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep,
snd_usbmidi_input_data(ep, 0, buffer, buffer_length);
}
-static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep)
+static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep,
+ struct urb *urb)
{
int count;
if (!ep->ports[0].active)
return;
- count = ep->urb->dev->speed == USB_SPEED_HIGH ? 1 : 2;
+ count = snd_usb_get_speed(ep->umidi->chip->dev) == USB_SPEED_HIGH
+ ? 1 : 2;
count = snd_rawmidi_transmit(ep->ports[0].substream,
- ep->urb->transfer_buffer,
+ urb->transfer_buffer,
count);
if (count < 1) {
ep->ports[0].active = 0;
return;
}
- memset(ep->urb->transfer_buffer + count, 0xFD, 9 - count);
- ep->urb->transfer_buffer_length = count;
+ memset(urb->transfer_buffer + count, 0xFD, 9 - count);
+ urb->transfer_buffer_length = count;
}
static struct usb_protocol_ops snd_usbmidi_122l_ops = {
@@ -786,10 +824,11 @@ static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep,
}
}
-static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep,
+ struct urb *urb)
{
int port0 = ep->current_port;
- uint8_t* buf = ep->urb->transfer_buffer;
+ uint8_t* buf = urb->transfer_buffer;
int buf_free = ep->max_transfer;
int length, i;
@@ -829,7 +868,7 @@ static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep)
*buf = 0xff;
--buf_free;
}
- ep->urb->transfer_buffer_length = ep->max_transfer - buf_free;
+ urb->transfer_buffer_length = ep->max_transfer - buf_free;
}
static struct usb_protocol_ops snd_usbmidi_emagic_ops = {
@@ -884,6 +923,35 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
}
}
+static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
+{
+ struct usbmidi_out_port* port = substream->runtime->private_data;
+ struct snd_usb_midi_out_endpoint *ep = port->ep;
+ unsigned int drain_urbs;
+ DEFINE_WAIT(wait);
+ long timeout = msecs_to_jiffies(50);
+
+ /*
+ * The substream buffer is empty, but some data might still be in the
+ * currently active URBs, so we have to wait for those to complete.
+ */
+ spin_lock_irq(&ep->buffer_lock);
+ drain_urbs = ep->active_urbs;
+ if (drain_urbs) {
+ ep->drain_urbs |= drain_urbs;
+ do {
+ prepare_to_wait(&ep->drain_wait, &wait,
+ TASK_UNINTERRUPTIBLE);
+ spin_unlock_irq(&ep->buffer_lock);
+ timeout = schedule_timeout(timeout);
+ spin_lock_irq(&ep->buffer_lock);
+ drain_urbs &= ep->drain_urbs;
+ } while (drain_urbs && timeout);
+ finish_wait(&ep->drain_wait, &wait);
+ }
+ spin_unlock_irq(&ep->buffer_lock);
+}
+
static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream)
{
return 0;
@@ -908,6 +976,7 @@ static struct snd_rawmidi_ops snd_usbmidi_output_ops = {
.open = snd_usbmidi_output_open,
.close = snd_usbmidi_output_close,
.trigger = snd_usbmidi_output_trigger,
+ .drain = snd_usbmidi_output_drain,
};
static struct snd_rawmidi_ops snd_usbmidi_input_ops = {
@@ -916,19 +985,26 @@ static struct snd_rawmidi_ops snd_usbmidi_input_ops = {
.trigger = snd_usbmidi_input_trigger
};
+static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb,
+ unsigned int buffer_length)
+{
+ usb_buffer_free(umidi->chip->dev, buffer_length,
+ urb->transfer_buffer, urb->transfer_dma);
+ usb_free_urb(urb);
+}
+
/*
* Frees an input endpoint.
* May be called when ep hasn't been initialized completely.
*/
static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep)
{
- if (ep->urb) {
- usb_buffer_free(ep->umidi->chip->dev,
- ep->urb->transfer_buffer_length,
- ep->urb->transfer_buffer,
- ep->urb->transfer_dma);
- usb_free_urb(ep->urb);
- }
+ unsigned int i;
+
+ for (i = 0; i < INPUT_URBS; ++i)
+ if (ep->urbs[i])
+ free_urb_and_buffer(ep->umidi, ep->urbs[i],
+ ep->urbs[i]->transfer_buffer_length);
kfree(ep);
}
@@ -943,6 +1019,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
void* buffer;
unsigned int pipe;
int length;
+ unsigned int i;
rep->in = NULL;
ep = kzalloc(sizeof(*ep), GFP_KERNEL);
@@ -950,30 +1027,36 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
return -ENOMEM;
ep->umidi = umidi;
- ep->urb = usb_alloc_urb(0, GFP_KERNEL);
- if (!ep->urb) {
- snd_usbmidi_in_endpoint_delete(ep);
- return -ENOMEM;
+ for (i = 0; i < INPUT_URBS; ++i) {
+ ep->urbs[i] = usb_alloc_urb(0, GFP_KERNEL);
+ if (!ep->urbs[i]) {
+ snd_usbmidi_in_endpoint_delete(ep);
+ return -ENOMEM;
+ }
}
if (ep_info->in_interval)
pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep);
else
pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep);
length = usb_maxpacket(umidi->chip->dev, pipe, 0);
- buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL,
- &ep->urb->transfer_dma);
- if (!buffer) {
- snd_usbmidi_in_endpoint_delete(ep);
- return -ENOMEM;
+ for (i = 0; i < INPUT_URBS; ++i) {
+ buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL,
+ &ep->urbs[i]->transfer_dma);
+ if (!buffer) {
+ snd_usbmidi_in_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->in_interval)
+ usb_fill_int_urb(ep->urbs[i], umidi->chip->dev,
+ pipe, buffer, length,
+ snd_usbmidi_in_urb_complete,
+ ep, ep_info->in_interval);
+ else
+ usb_fill_bulk_urb(ep->urbs[i], umidi->chip->dev,
+ pipe, buffer, length,
+ snd_usbmidi_in_urb_complete, ep);
+ ep->urbs[i]->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
}
- if (ep_info->in_interval)
- usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer,
- length, snd_usbmidi_in_urb_complete, ep,
- ep_info->in_interval);
- else
- usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer,
- length, snd_usbmidi_in_urb_complete, ep);
- ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
rep->in = ep;
return 0;
@@ -994,12 +1077,12 @@ static unsigned int snd_usbmidi_count_bits(unsigned int x)
*/
static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep)
{
- if (ep->urb) {
- usb_buffer_free(ep->umidi->chip->dev, ep->max_transfer,
- ep->urb->transfer_buffer,
- ep->urb->transfer_dma);
- usb_free_urb(ep->urb);
- }
+ unsigned int i;
+
+ for (i = 0; i < OUTPUT_URBS; ++i)
+ if (ep->urbs[i].urb)
+ free_urb_and_buffer(ep->umidi, ep->urbs[i].urb,
+ ep->max_transfer);
kfree(ep);
}
@@ -1011,7 +1094,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
struct snd_usb_midi_endpoint* rep)
{
struct snd_usb_midi_out_endpoint* ep;
- int i;
+ unsigned int i;
unsigned int pipe;
void* buffer;
@@ -1021,38 +1104,46 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
return -ENOMEM;
ep->umidi = umidi;
- ep->urb = usb_alloc_urb(0, GFP_KERNEL);
- if (!ep->urb) {
- snd_usbmidi_out_endpoint_delete(ep);
- return -ENOMEM;
+ for (i = 0; i < OUTPUT_URBS; ++i) {
+ ep->urbs[i].urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!ep->urbs[i].urb) {
+ snd_usbmidi_out_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ ep->urbs[i].ep = ep;
}
if (ep_info->out_interval)
pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep);
else
pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep);
if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */
- /* FIXME: we need more URBs to get reasonable bandwidth here: */
ep->max_transfer = 4;
else
ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1);
- buffer = usb_buffer_alloc(umidi->chip->dev, ep->max_transfer,
- GFP_KERNEL, &ep->urb->transfer_dma);
- if (!buffer) {
- snd_usbmidi_out_endpoint_delete(ep);
- return -ENOMEM;
+ for (i = 0; i < OUTPUT_URBS; ++i) {
+ buffer = usb_buffer_alloc(umidi->chip->dev,
+ ep->max_transfer, GFP_KERNEL,
+ &ep->urbs[i].urb->transfer_dma);
+ if (!buffer) {
+ snd_usbmidi_out_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->out_interval)
+ usb_fill_int_urb(ep->urbs[i].urb, umidi->chip->dev,
+ pipe, buffer, ep->max_transfer,
+ snd_usbmidi_out_urb_complete,
+ &ep->urbs[i], ep_info->out_interval);
+ else
+ usb_fill_bulk_urb(ep->urbs[i].urb, umidi->chip->dev,
+ pipe, buffer, ep->max_transfer,
+ snd_usbmidi_out_urb_complete,
+ &ep->urbs[i]);
+ ep->urbs[i].urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
}
- if (ep_info->out_interval)
- usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer,
- ep->max_transfer, snd_usbmidi_out_urb_complete,
- ep, ep_info->out_interval);
- else
- usb_fill_bulk_urb(ep->urb, umidi->chip->dev,
- pipe, buffer, ep->max_transfer,
- snd_usbmidi_out_urb_complete, ep);
- ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
spin_lock_init(&ep->buffer_lock);
tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep);
+ init_waitqueue_head(&ep->drain_wait);
for (i = 0; i < 0x10; ++i)
if (ep_info->out_cables & (1 << i)) {
@@ -1090,7 +1181,7 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi)
void snd_usbmidi_disconnect(struct list_head* p)
{
struct snd_usb_midi* umidi;
- int i;
+ unsigned int i, j;
umidi = list_entry(p, struct snd_usb_midi, list);
/*
@@ -1105,13 +1196,15 @@ void snd_usbmidi_disconnect(struct list_head* p)
struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
if (ep->out)
tasklet_kill(&ep->out->tasklet);
- if (ep->out && ep->out->urb) {
- usb_kill_urb(ep->out->urb);
+ if (ep->out) {
+ for (j = 0; j < OUTPUT_URBS; ++j)
+ usb_kill_urb(ep->out->urbs[j].urb);
if (umidi->usb_protocol_ops->finish_out_endpoint)
umidi->usb_protocol_ops->finish_out_endpoint(ep->out);
}
if (ep->in)
- usb_kill_urb(ep->in->urb);
+ for (j = 0; j < INPUT_URBS; ++j)
+ usb_kill_urb(ep->in->urbs[j]);
/* free endpoints here; later call can result in Oops */
if (ep->out) {
snd_usbmidi_out_endpoint_delete(ep->out);
@@ -1692,20 +1785,25 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi,
void snd_usbmidi_input_stop(struct list_head* p)
{
struct snd_usb_midi* umidi;
- int i;
+ unsigned int i, j;
umidi = list_entry(p, struct snd_usb_midi, list);
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
if (ep->in)
- usb_kill_urb(ep->in->urb);
+ for (j = 0; j < INPUT_URBS; ++j)
+ usb_kill_urb(ep->in->urbs[j]);
}
}
static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep)
{
- if (ep) {
- struct urb* urb = ep->urb;
+ unsigned int i;
+
+ if (!ep)
+ return;
+ for (i = 0; i < INPUT_URBS; ++i) {
+ struct urb* urb = ep->urbs[i];
urb->dev = ep->umidi->chip->dev;
snd_usbmidi_submit_urb(urb, GFP_KERNEL);
}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ec9cdf9..ab5a3ac 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -86,6 +86,7 @@ struct usb_mixer_interface {
u8 rc_buffer[6];
u8 audigy2nx_leds[3];
+ u8 xonar_u1_status;
};
@@ -461,7 +462,7 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *_tlv)
{
struct usb_mixer_elem_info *cval = kcontrol->private_data;
- DECLARE_TLV_DB_SCALE(scale, 0, 0, 0);
+ DECLARE_TLV_DB_MINMAX(scale, 0, 0);
if (size < sizeof(scale))
return -ENOMEM;
@@ -469,7 +470,16 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
* while ALSA TLV contains in 1/100 dB unit
*/
scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256;
- scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256;
+ scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256;
+ if (scale[3] <= scale[2]) {
+ /* something is wrong; assume it's either from/to 0dB */
+ if (scale[2] < 0)
+ scale[3] = 0;
+ else if (scale[2] > 0)
+ scale[2] = 0;
+ else /* totally crap, return an error */
+ return -EINVAL;
+ }
if (copy_to_user(_tlv, scale, sizeof(scale)))
return -EFAULT;
return 0;
@@ -2033,6 +2043,58 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
}
}
+static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02);
+ return 0;
+}
+
+static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ u8 old_status, new_status;
+ int err, changed;
+
+ old_status = mixer->xonar_u1_status;
+ if (ucontrol->value.integer.value[0])
+ new_status = old_status | 0x02;
+ else
+ new_status = old_status & ~0x02;
+ changed = new_status != old_status;
+ err = snd_usb_ctl_msg(mixer->chip->dev,
+ usb_sndctrlpipe(mixer->chip->dev, 0), 0x08,
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ 50, 0, &new_status, 1, 100);
+ if (err < 0)
+ return err;
+ mixer->xonar_u1_status = new_status;
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_xonar_u1_output_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = snd_xonar_u1_switch_get,
+ .put = snd_xonar_u1_switch_put,
+};
+
+static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer)
+{
+ int err;
+
+ err = snd_ctl_add(mixer->chip->card,
+ snd_ctl_new1(&snd_xonar_u1_output_switch, mixer));
+ if (err < 0)
+ return err;
+ mixer->xonar_u1_status = 0x05;
+ return 0;
+}
+
int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
int ignore_error)
{
@@ -2075,6 +2137,13 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
snd_audigy2nx_proc_read);
}
+ if (mixer->chip->usb_id == USB_ID(0x0b05, 0x1739) ||
+ mixer->chip->usb_id == USB_ID(0x0b05, 0x1743)) {
+ err = snd_xonar_u1_controls_create(mixer);
+ if (err < 0)
+ goto _error;
+ }
+
err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops);
if (err < 0)
goto _error;
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