diff options
Diffstat (limited to 'Documentation/sound/alsa')
28 files changed, 13192 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt new file mode 100644 index 0000000..bef6550 --- /dev/null +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -0,0 +1,2417 @@ + + Advanced Linux Sound Architecture - Driver + ========================================== + Configuration guide + + +Kernel Configuration +==================== + +To enable ALSA support you need at least to build the kernel with +primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS, +you don't have to choose any of the OSS modules. + +Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and +PCM supports if you want to run OSS applications with ALSA. + +If you want to support the WaveTable functionality on cards such as +SB Live! then you need to enable "Sequencer support" +(CONFIG_SND_SEQUENCER). + +To make ALSA debug messages more verbose, enable the "Verbose printk" +and "Debug" options. To check for memory leaks, turn on "Debug memory" +too. "Debug detection" will add checks for the detection of cards. + +Please note that all the ALSA ISA drivers support the Linux isapnp API +(if the card supports ISA PnP). You don't need to configure the cards +using isapnptools. + + +Creating ALSA devices +===================== + +This depends on your distribution, but normally you use the /dev/MAKEDEV +script to create the necessary device nodes. On some systems you use a +script named 'snddevices'. + + +Module parameters +================= + +The user can load modules with options. If the module supports more than +one card and you have more than one card of the same type then you can +specify multiple values for the option separated by commas. + +Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. + + Module snd + ---------- + + The core ALSA module. It is used by all ALSA card drivers. + It takes the following options which have global effects. + + major - major number for sound driver + - Default: 116 + cards_limit + - limiting card index for auto-loading (1-8) + - Default: 1 + - For auto-loading more than one card, specify this + option together with snd-card-X aliases. + slots - Reserve the slot index for the given driver. + This option takes multiple strings. + See "Module Autoloading Support" section for details. + + Module snd-pcm-oss + ------------------ + + The PCM OSS emulation module. + This module takes options which change the mapping of devices. + + dsp_map - PCM device number maps assigned to the 1st OSS device. + - Default: 0 + adsp_map - PCM device number maps assigned to the 2st OSS device. + - Default: 1 + nonblock_open + - Don't block opening busy PCM devices. Default: 1 + + For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of + the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped + to PCM #0 of the card #0. + For changing the second or later card, specify the option with + commas, such like "dsp_map=0,1". + + nonblock_open option is used to change the behavior of the PCM + regarding opening the device. When this option is non-zero, + opening a busy OSS PCM device won't be blocked but return + immediately with EAGAIN (just like O_NONBLOCK flag). + + Module snd-rawmidi + ------------------ + + This module takes options which change the mapping of devices. + similar to those of the snd-pcm-oss module. + + midi_map - MIDI device number maps assigned to the 1st OSS device. + - Default: 0 + amidi_map - MIDI device number maps assigned to the 2st OSS device. + - Default: 1 + + Common parameters for top sound card modules + -------------------------------------------- + + Each of top level sound card module takes the following options. + + index - index (slot #) of sound card + - Values: 0 through 31 or negative + - If nonnegative, assign that index number + - if negative, interpret as a bitmask of permissible + indices; the first free permitted index is assigned + - Default: -1 + id - card ID (identifier or name) + - Can be up to 15 characters long + - Default: the card type + - A directory by this name is created under /proc/asound/ + containing information about the card + - This ID can be used instead of the index number in + identifying the card + enable - enable card + - Default: enabled, for PCI and ISA PnP cards + + Module snd-adlib + ---------------- + + Module for AdLib FM cards. + + port - port # for OPL chip + + This module supports multiple cards. It does not support autoprobe, so + the port must be specified. For actual AdLib FM cards it will be 0x388. + Note that this card does not have PCM support and no mixer; only FM + synthesis. + + Make sure you have "sbiload" from the alsa-tools package available and, + after loading the module, find out the assigned ALSA sequencer port + number through "sbiload -l". Example output: + + Port Client name Port name + 64:0 OPL2 FM synth OPL2 FM Port + + Load the std.sb and drums.sb patches also supplied by sbiload: + + sbiload -p 64:0 std.sb drums.sb + + If you use this driver to drive an OPL3, you can use std.o3 and drums.o3 + instead. To have the card produce sound, use aplaymidi from alsa-utils: + + aplaymidi -p 64:0 foo.mid + + Module snd-ad1816a + ------------------ + + Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. + + clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) + + This module supports multiple cards, autoprobe and PnP. + + Module snd-ad1848 + ----------------- + + Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. + + port - port # for AD1848 chip + irq - IRQ # for AD1848 chip + dma1 - DMA # for AD1848 chip (0,1,3) + + This module supports multiple cards. It does not support autoprobe + thus main port must be specified!!! Other ports are optional. + + The power-management is supported. + + Module snd-ad1889 + ----------------- + + Module for Analog Devices AD1889 chips. + + ac97_quirk - AC'97 workaround for strange hardware + See the description of intel8x0 module for details. + + This module supports multiple cards. + + Module snd-ali5451 + ------------------ + + Module for ALi M5451 PCI chip. + + pcm_channels - Number of hardware channels assigned for PCM + spdif - Support SPDIF I/O + - Default: disabled + + This module supports one chip and autoprobe. + + The power-management is supported. + + Module snd-als100 + ----------------- + + Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. + + This module supports multiple cards, autoprobe and PnP. + + The power-management is supported. + + Module snd-als300 + ----------------- + + Module for Avance Logic ALS300 and ALS300+ + + This module supports multiple cards. + + The power-management is supported. + + Module snd-als4000 + ------------------ + + Module for sound cards based on Avance Logic ALS4000 PCI chip. + + joystick_port - port # for legacy joystick support. + 0 = disabled (default), 1 = auto-detect + + This module supports multiple cards, autoprobe and PnP. + + The power-management is supported. + + Module snd-atiixp + ----------------- + + Module for ATI IXP 150/200/250/400 AC97 controllers. + + ac97_clock - AC'97 clock (default = 48000) + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + ac97_codec - Workaround to specify which AC'97 codec + instead of probing. If this works for you + file a bug with your `lspci -vn` output. + -2 -- Force probing. + -1 -- Default behavior. + 0-2 -- Use the specified codec. + spdif_aclink - S/PDIF transfer over AC-link (default = 1) + + This module supports one card and autoprobe. + + ATI IXP has two different methods to control SPDIF output. One is + over AC-link and another is over the "direct" SPDIF output. The + implementation depends on the motherboard, and you'll need to + choose the correct one via spdif_aclink module option. + + The power-management is supported. + + Module snd-atiixp-modem + ----------------------- + + Module for ATI IXP 150/200/250 AC97 modem controllers. + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-au8810, snd-au8820, snd-au8830 + ----------------------------------------- + + Module for Aureal Vortex, Vortex2 and Advantage device. + + pcifix - Control PCI workarounds + 0 = Disable all workarounds + 1 = Force the PCI latency of the Aureal card to 0xff + 2 = Force the Extend PCI#2 Internal Master for Efficient + Handling of Dummy Requests on the VIA KT133 AGP Bridge + 3 = Force both settings + 255 = Autodetect what is required (default) + + This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware + EQ, mpu401, gameport. A3D and wavetable support are still in development. + Development and reverse engineering work is being coordinated at + http://savannah.nongnu.org/projects/openvortex/ + SPDIF output has a copy of the AC97 codec output, unless you use the + "spdif" pcm device, which allows raw data passthru. + The hardware EQ hardware and SPDIF is only present in the Vortex2 and + Advantage. + + Note: Some ALSA mixer applications don't handle the SPDIF sample rate + control correctly. If you have problems regarding this, try + another ALSA compliant mixer (alsamixer works). + + Module snd-aw2 + -------------- + + Module for Audiowerk2 sound card + + This module supports multiple cards. + + Module snd-azt2320 + ------------------ + + Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). + + This module supports multiple cards, PnP and autoprobe. + + The power-management is supported. + + Module snd-azt3328 + ------------------ + + Module for sound cards based on Aztech AZF3328 PCI chip. + + joystick - Enable joystick (default off) + + This module supports multiple cards. + + Module snd-bt87x + ---------------- + + Module for video cards based on Bt87x chips. + + digital_rate - Override the default digital rate (Hz) + load_all - Load the driver even if the card model isn't known + + This module supports multiple cards. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + Module snd-ca0106 + ----------------- + + Module for Creative Audigy LS and SB Live 24bit + + This module supports multiple cards. + + + Module snd-cmi8330 + ------------------ + + Module for sound cards based on C-Media CMI8330 ISA chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + wssport - port # for CMI8330 chip (WSS) + wssirq - IRQ # for CMI8330 chip (WSS) + wssdma - first DMA # for CMI8330 chip (WSS) + sbport - port # for CMI8330 chip (SB16) + sbirq - IRQ # for CMI8330 chip (SB16) + sbdma8 - 8bit DMA # for CMI8330 chip (SB16) + sbdma16 - 16bit DMA # for CMI8330 chip (SB16) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-cmipci + ----------------- + + Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. + + mpu_port - port address of MIDI interface (8338 only): + 0x300,0x310,0x320,0x330 = legacy port, + 0 = disable (default) + fm_port - port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), + 0 = disable + soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) + (default = 1) + joystick_port - Joystick port address (0 = disable, 1 = auto-detect) + + This module supports autoprobe and multiple cards. + + The power-management is supported. + + Module snd-cs4231 + ----------------- + + Module for sound cards based on CS4231 ISA chips. + + port - port # for CS4231 chip + mpu_port - port # for MPU-401 UART (optional), -1 = disable + irq - IRQ # for CS4231 chip + mpu_irq - IRQ # for MPU-401 UART + dma1 - first DMA # for CS4231 chip + dma2 - second DMA # for CS4231 chip + + This module supports multiple cards. This module does not support autoprobe + thus main port must be specified!!! Other ports are optional. + + The power-management is supported. + + Module snd-cs4232 + ----------------- + + Module for sound cards based on CS4232/CS4232A ISA chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for CS4232 chip (PnP setup - 0x534) + cport - control port # for CS4232 chip (PnP setup - 0x120,0x210,0xf00) + mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable + fm_port - FM port # for CS4232 chip (PnP setup - 0x388), -1 = disable + irq - IRQ # for CS4232 chip (5,7,9,11,12,15) + mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) + dma1 - first DMA # for CS4232 chip (0,1,3) + dma2 - second DMA # for Yamaha CS4232 chip (0,1,3), -1 = disable + + This module supports multiple cards. This module does not support autoprobe + (if ISA PnP is not used) thus main port must be specified!!! Other ports are + optional. + + The power-management is supported. + + Module snd-cs4236 + ----------------- + + Module for sound cards based on CS4235/CS4236/CS4236B/CS4237B/ + CS4238B/CS4239 ISA chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for CS4236 chip (PnP setup - 0x534) + cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) + mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable + fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable + irq - IRQ # for CS4236 chip (5,7,9,11,12,15) + mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) + dma1 - first DMA # for CS4236 chip (0,1,3) + dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable + + This module supports multiple cards. This module does not support autoprobe + (if ISA PnP is not used) thus main port and control port must be + specified!!! Other ports are optional. + + The power-management is supported. + + Module snd-cs4281 + ----------------- + + Module for Cirrus Logic CS4281 soundchip. + + dual_codec - Secondary codec ID (0 = disable, default) + + This module supports multiple cards. + + The power-management is supported. + + Module snd-cs46xx + ----------------- + + Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ + CS4624/CS4630/CS4280 PCI chips. + + external_amp - Force to enable external amplifier. + thinkpad - Force to enable Thinkpad's CLKRUN control. + mmap_valid - Support OSS mmap mode (default = 0). + + This module supports multiple cards and autoprobe. + Usually external amp and CLKRUN controls are detected automatically + from PCI sub vendor/device ids. If they don't work, give the options + above explicitly. + + The power-management is supported. + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + + Module snd-cs5535audio + ---------------------- + + Module for multifunction CS5535 companion PCI device + + The power-management is supported. + + Module snd-darla20 + ------------------ + + Module for Echoaudio Darla20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-darla24 + ------------------ + + Module for Echoaudio Darla24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-dt019x + ----------------- + + Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP + only) + + This module supports multiple cards. This module is enabled only with + ISA PnP support. + + The power-management is supported. + + Module snd-dummy + ---------------- + + Module for the dummy sound card. This "card" doesn't do any output + or input, but you may use this module for any application which + requires a sound card (like RealPlayer). + + The power-management is supported. + + Module snd-echo3g + ----------------- + + Module for Echoaudio 3G cards (Gina3G/Layla3G) + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-emu10k1 + ------------------ + + Module for EMU10K1/EMU10k2 based PCI sound cards. + * Sound Blaster Live! + * Sound Blaster PCI 512 + * Emu APS (partially supported) + * Sound Blaster Audigy + + extin - bitmap of available external inputs for FX8010 (see bellow) + extout - bitmap of available external outputs for FX8010 (see bellow) + seq_ports - allocated sequencer ports (4 by default) + max_synth_voices - limit of voices used for wavetable (64 by default) + max_buffer_size - specifies the maximum size of wavetable/pcm buffers + given in MB unit. Default value is 128. + enable_ir - enable IR + + This module supports multiple cards and autoprobe. + + Input & Output configurations [extin/extout] + * Creative Card wo/Digital out [0x0003/0x1f03] + * Creative Card w/Digital out [0x0003/0x1f0f] + * Creative Card w/Digital CD in [0x000f/0x1f0f] + * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] + * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] + * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] + * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] + * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] + * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] + * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] + * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] + * Creative Card all ins and outs [0x3fff/0x7fff] + + The power-management is supported. + + Module snd-emu10k1x + ------------------- + + Module for Creative Emu10k1X (SB Live Dell OEM version) + + This module supports multiple cards. + + Module snd-ens1370 + ------------------ + + Module for Ensoniq AudioPCI ES1370 PCI sound cards. + * SoundBlaster PCI 64 + * SoundBlaster PCI 128 + + joystick - Enable joystick (default off) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-ens1371 + ------------------ + + Module for Ensoniq AudioPCI ES1371 PCI sound cards. + * SoundBlaster PCI 64 + * SoundBlaster PCI 128 + * SoundBlaster Vibra PCI + + joystick_port - port # for joystick (0x200,0x208,0x210,0x218), + 0 = disable (default), 1 = auto-detect + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-es968 + ---------------- + + Module for sound cards based on ESS ES968 chip (PnP only). + + This module supports multiple cards, PnP and autoprobe. + + The power-management is supported. + + Module snd-es1688 + ----------------- + + Module for ESS AudioDrive ES-1688 and ES-688 sound cards. + + port - port # for ES-1688 chip (0x220,0x240,0x260) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) + irq - IRQ # for ES-1688 chip (5,7,9,10) + mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + dma8 - DMA # for ES-1688 chip (0,1,3) + + This module supports multiple cards and autoprobe (without MPU-401 port). + + Module snd-es18xx + ----------------- + + Module for ESS AudioDrive ES-18xx sound cards. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for ES-18xx chip (0x220,0x240,0x260) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) + fm_port - port # for FM (optional, not used) + irq - IRQ # for ES-18xx chip (5,7,9,10) + dma1 - first DMA # for ES-18xx chip (0,1,3) + dma2 - first DMA # for ES-18xx chip (0,1,3) + + This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 + port if native ISA PnP routines are not used). + When dma2 is equal with dma1, the driver works as half-duplex. + + The power-management is supported. + + Module snd-es1938 + ----------------- + + Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-es1968 + ----------------- + + Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. + + total_bufsize - total buffer size in kB (1-4096kB) + pcm_substreams_p - playback channels (1-8, default=2) + pcm_substreams_c - capture channels (1-8, default=0) + clock - clock (0 = auto-detection) + use_pm - support the power-management (0 = off, 1 = on, + 2 = auto (default)) + enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default)) + joystick - enable joystick (default off) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-fm801 + ---------------- + + Module for ForteMedia FM801 based PCI sound cards. + + tea575x_tuner - Enable TEA575x tuner + - 1 = MediaForte 256-PCS + - 2 = MediaForte 256-PCPR + - 3 = MediaForte 64-PCR + - High 16-bits are video (radio) device number + 1 + - example: 0x10002 (MediaForte 256-PCPR, device 1) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-gina20 + ----------------- + + Module for Echoaudio Gina20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-gina24 + ----------------- + + Module for Echoaudio Gina24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-gusclassic + --------------------- + + Module for Gravis UltraSound Classic sound card. + + port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for GF1 chip (3,5,9,11,12,15) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe. + + Module snd-gusextreme + --------------------- + + Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. + + port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) + gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable + irq - IRQ # for ES-1688 chip (5,7,9,10) + gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15) + mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + dma8 - DMA # for ES-1688 chip (0,1,3) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe (without MPU-401 port). + + Module snd-gusmax + ----------------- + + Module for Gravis UltraSound MAX sound card. + + port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for GF1 chip (3,5,9,11,12,15) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe. + + Module snd-hda-intel + -------------------- + + Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, + PCH, SCH), + ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, + RV630, RV635, RV670, RV770, + VIA VT8251/VT8237A, + SIS966, ULI M5461 + + [Multiple options for each card instance] + model - force the model name + position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) + probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + 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. + + [Single (global) options] + single_cmd - Use single immediate commands to communicate with + codecs (for debugging only) + enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) + power_save - Automatic power-saving timtout (in second, 0 = + disable) + power_save_controller - Reset HD-audio controller in power-saving mode + (default = on) + + This module supports multiple cards and autoprobe. + + Each codec may have a model table for different configurations. + If your machine isn't listed there, the default (usually minimal) + configuration is set up. You can pass "model=<name>" option to + specify a certain model in such a case. There are different + models depending on the codec chip. + + Model name Description + ---------- ----------- + ALC880 + 3stack 3-jack in back and a headphone out + 3stack-digout 3-jack in back, a HP out and a SPDIF out + 5stack 5-jack in back, 2-jack in front + 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out + 6stack 6-jack in back, 2-jack in front + 6stack-digout 6-jack with a SPDIF out + w810 3-jack + z71v 3-jack (HP shared SPDIF) + asus 3-jack (ASUS Mobo) + asus-w1v ASUS W1V + asus-dig ASUS with SPDIF out + asus-dig2 ASUS with SPDIF out (using GPIO2) + uniwill 3-jack + fujitsu Fujitsu Laptops (Pi1536) + F1734 2-jack + lg LG laptop (m1 express dual) + lg-lw LG LW20/LW25 laptop + tcl TCL S700 + clevo Clevo laptops (m520G, m665n) + medion Medion Rim 2150 + test for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + + ALC260 + hp HP machines + hp-3013 HP machines (3013-variant) + hp-dc7600 HP DC7600 + fujitsu Fujitsu S7020 + acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V + basic fixed pin assignment (old default model) + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + + ALC262 + fujitsu Fujitsu Laptop + hp-bpc HP xw4400/6400/8400/9400 laptops + hp-bpc-d7000 HP BPC D7000 + hp-tc-t5735 HP Thin Client T5735 + hp-rp5700 HP RP5700 + benq Benq ED8 + benq-t31 Benq T31 + hippo Hippo (ATI) with jack detection, Sony UX-90s + hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD + toshiba-s06 Toshiba S06 + toshiba-rx1 Toshiba RX1 + ultra Samsung Q1 Ultra Vista model + lenovo-3000 Lenovo 3000 y410 + nec NEC Versa S9100 + basic fixed pin assignment w/o SPDIF + auto auto-config reading BIOS (default) + + ALC267/268 + quanta-il1 Quanta IL1 mini-notebook + 3stack 3-stack model + toshiba Toshiba A205 + acer Acer laptops + acer-aspire Acer Aspire One + dell Dell OEM laptops (Vostro 1200) + zepto Zepto laptops + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + + ALC269 + basic Basic preset + quanta Quanta FL1 + eeepc-p703 ASUS Eeepc P703 P900A + eeepc-p901 ASUS Eeepc P901 S101 + + ALC662/663 + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 6stack-dig 6-stack with SPDIF + lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc P701 + eeepc-ep20 ASUS Eeepc EP20 + ecs ECS/Foxconn mobo + m51va ASUS M51VA + g71v ASUS G71V + h13 ASUS H13 + g50v ASUS G50V + asus-mode1 ASUS + asus-mode2 ASUS + asus-mode3 ASUS + asus-mode4 ASUS + asus-mode5 ASUS + asus-mode6 ASUS + auto auto-config reading BIOS (default) + + ALC882/885 + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O + arima Arima W820Di1 + targa Targa T8, MSI-1049 T8 + asus-a7j ASUS A7J + asus-a7m ASUS A7M + macpro MacPro support + 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-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 + acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 + medion Medion Laptops + medion-md2 Medion MD2 + targa-dig Targa/MSI + targa-2ch-dig Targs/MSI with 2-channel + laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + lenovo-sky Lenovo Sky + haier-w66 Haier W66 + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + 6stack-dell Dell machines with 6stack (Inspiron 530) + mitac Mitac 8252D + clevo-m720 Clevo M720 laptop series + fujitsu-pi2515 Fujitsu AMILO Pi2515 + 3stack-6ch-intel Intel DG33* boards + auto auto-config reading BIOS (default) + + ALC861/660 + 3stack 3-jack + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack with SPDIF I/O + 3stack-660 3-jack (for ALC660) + uniwill-m31 Uniwill M31 laptop + toshiba Toshiba laptop support + asus Asus laptop support + asus-laptop ASUS F2/F3 laptops + auto auto-config reading BIOS (default) + + ALC861VD/660VD + 3stack 3-jack + 3stack-dig 3-jack with SPDIF OUT + 6stack-dig 6-jack with SPDIF OUT + 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) + lenovo Lenovo 3000 C200 + dallas Dallas laptops + hp HP TX1000 + auto auto-config reading BIOS (default) + + CMI9880 + minimal 3-jack in back + min_fp 3-jack in back, 2-jack in front + full 6-jack in back, 2-jack in front + full_dig 6-jack in back, 2-jack in front, SPDIF I/O + allout 5-jack in back, 2-jack in front, SPDIF out + auto auto-config reading BIOS (default) + + AD1882 / AD1882A + 3stack 3-stack mode (default) + 6stack 6-stack mode + + AD1884A / AD1883 / AD1984A / AD1984B + desktop 3-stack desktop (default) + laptop laptop with HP jack sensing + mobile mobile devices with HP jack sensing + thinkpad Lenovo Thinkpad X300 + + AD1884 + N/A + + AD1981 + basic 3-jack (default) + hp HP nx6320 + thinkpad Lenovo Thinkpad T60/X60/Z60 + toshiba Toshiba U205 + + AD1983 + N/A + + AD1984 + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + dell Dell T3400 + + AD1986A + 6stack 6-jack, separate surrounds (default) + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + laptop-eapd 2-channel with EAPD (ASUS A6J) + laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) + ultra 2-channel with EAPD (Samsung Ultra tablet PC) + samsung 2-channel with EAPD (Samsung R65) + + AD1988/AD1988B/AD1989A/AD1989B + 6stack 6-jack + 6stack-dig ditto with SPDIF + 3stack 3-jack + 3stack-dig ditto with SPDIF + laptop 3-jack with hp-jack automute + laptop-dig ditto with SPDIF + auto auto-config reading BIOS (default) + + Conexant 5045 + laptop-hpsense Laptop with HP sense (old model laptop) + laptop-micsense Laptop with Mic sense (old model fujitsu) + laptop-hpmicsense Laptop with HP and Mic senses + benq Benq R55E + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + + Conexant 5047 + laptop Basic Laptop config + laptop-hp Laptop config for some HP models (subdevice 30A5) + laptop-eapd Laptop config with EAPD support + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + + Conexant 5051 + laptop Basic Laptop config (default) + hp HP Spartan laptop + + STAC9200 + ref Reference board + dell-d21 Dell (unknown) + dell-d22 Dell (unknown) + dell-d23 Dell (unknown) + dell-m21 Dell Inspiron 630m, Dell Inspiron 640m + dell-m22 Dell Latitude D620, Dell Latitude D820 + dell-m23 Dell XPS M1710, Dell Precision M90 + dell-m24 Dell Latitude 120L + dell-m25 Dell Inspiron E1505n + dell-m26 Dell Inspiron 1501 + dell-m27 Dell Inspiron E1705/9400 + gateway Gateway laptops with EAPD control + panasonic Panasonic CF-74 + + STAC9205/9254 + ref Reference board + dell-m42 Dell (unknown) + dell-m43 Dell Precision + dell-m44 Dell Inspiron + + STAC9220/9221 + ref Reference board + 3stack D945 3stack + 5stack D945 5stack + SPDIF + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + intel-mac-auto Intel Mac (detect type according to subsystem id) + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) + dell-d81 Dell (unknown) + dell-d82 Dell (unknown) + dell-m81 Dell (unknown) + dell-m82 Dell XPS M1210 + + STAC9202/9250/9251 + ref Reference board, base config + m2-2 Some Gateway MX series laptops + m6 Some Gateway NX series laptops + pa6 Gateway NX860 series + + STAC9227/9228/9229/927x + ref Reference board + ref-no-jd Reference board without HP/Mic jack detection + 3stack D965 3stack + 5stack D965 5stack + SPDIF + dell-3stack Dell Dimension E520 + dell-bios Fixes with Dell BIOS setup + + STAC92HD71B* + ref Reference board + dell-m4-1 Dell desktops + dell-m4-2 Dell desktops + dell-m4-3 Dell desktops + + STAC92HD73* + ref Reference board + no-jd BIOS setup but without jack-detection + 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 + + STAC9872 + vaio Setup for VAIO FE550G/SZ110 + vaio-ar Setup for VAIO AR + + The model name "genric" is treated as a special case. When this + model is given, the driver uses the generic codec parser without + "codec-patch". It's sometimes good for testing and debugging. + + If the default configuration doesn't work and one of the above + matches with your device, report it together with the PCI + subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel + ML (see the section "Links and Addresses"). + + power_save and power_save_controller options are for power-saving + mode. See powersave.txt for details. + + Note 2: If you get click noises on output, try the module option + position_fix=1 or 2. position_fix=1 will use the SD_LPIB + register value without FIFO size correction as the current + DMA pointer. position_fix=2 will make the driver to use + the position buffer instead of reading SD_LPIB register. + (Usually SD_LPIB register is more accurate than the + position buffer.) + + NB: If you get many "azx_get_response timeout" messages at + loading, it's likely a problem of interrupts (e.g. ACPI irq + routing). Try to boot with options like "pci=noacpi". Also, you + can try "single_cmd=1" module option. This will switch the + communication method between HDA controller and codecs to the + single immediate commands instead of CORB/RIRB. Basically, the + single command mode is provided only for BIOS, and you won't get + unsolicited events, too. But, at least, this works independently + from the irq. Remember this is a last resort, and should be + avoided as much as possible... + + MORE NOTES ON "azx_get_response timeout" PROBLEMS: + On some hardwares, you may need to add a proper probe_mask option + to avoid the "azx_get_response timeout" problem above, instead. + This occurs when the access to non-existing or non-working codec slot + (likely a modem one) causes a stall of the communication via HD-audio + bus. You can see which codec slots are probed by enabling + CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec + proc files. Then limit the slots to probe by probe_mask option. + For example, probe_mask=1 means to probe only the first slot, and + probe_mask=4 means only the third slot. + + The power-management is supported. + + Module snd-hdsp + --------------- + + Module for RME Hammerfall DSP audio interface(s) + + This module supports multiple cards. + + Note: The firmware data can be automatically loaded via hotplug + when CONFIG_FW_LOADER is set. Otherwise, you need to load + the firmware via hdsploader utility included in alsa-tools + package. + The firmware data is found in alsa-firmware package. + + Note: snd-page-alloc module does the job which snd-hammerfall-mem + module did formerly. It will allocate the buffers in advance + when any HDSP cards are found. To make the buffer + allocation sure, load snd-page-alloc module in the early + stage of boot sequence. See "Early Buffer Allocation" + section. + + Module snd-hdspm + ---------------- + + Module for RME HDSP MADI board. + + precise_ptr - Enable precise pointer, or disable. + line_outs_monitor - Send playback streams to analog outs by default. + enable_monitor - Enable Analog Out on Channel 63/64 by default. + + See hdspm.txt for details. + + Module snd-hifier + ----------------- + + Module for the MediaTek/TempoTec HiFier Fantasia sound card. + + This module supports autoprobe and multiple cards. + + Module snd-ice1712 + ------------------ + + Module for Envy24 (ICE1712) based PCI sound cards. + * MidiMan M Audio Delta 1010 + * MidiMan M Audio Delta 1010LT + * MidiMan M Audio Delta DiO 2496 + * MidiMan M Audio Delta 66 + * MidiMan M Audio Delta 44 + * MidiMan M Audio Delta 410 + * MidiMan M Audio Audiophile 2496 + * TerraTec EWS 88MT + * TerraTec EWS 88D + * TerraTec EWX 24/96 + * TerraTec DMX 6Fire + * TerraTec Phase 88 + * Hoontech SoundTrack DSP 24 + * Hoontech SoundTrack DSP 24 Value + * Hoontech SoundTrack DSP 24 Media 7.1 + * Event Electronics, EZ8 + * Digigram VX442 + * Lionstracs, Mediastaton + * Terrasoniq TS 88 + + model - Use the given board model, one of the following: + delta1010, dio2496, delta66, delta44, audiophile, delta410, + delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, + dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, + phase88, mediastation + omni - Omni I/O support for MidiMan M-Audio Delta44/66 + cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver) + in msec resolution, default value is 500 (0.5 sec) + + This module supports multiple cards and autoprobe. Note: The consumer part + is not used with all Envy24 based cards (for example in the MidiMan Delta + serie). + + Note: The supported board is detected by reading EEPROM or PCI + SSID (if EEPROM isn't available). You can override the + model by passing "model" module option in case that the + driver isn't configured properly or you want to try another + type for testing. + + Module snd-ice1724 + ------------------ + + Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. + * MidiMan M Audio Revolution 5.1 + * MidiMan M Audio Revolution 7.1 + * MidiMan M Audio Audiophile 192 + * AMP Ltd AUDIO2000 + * TerraTec Aureon 5.1 Sky + * TerraTec Aureon 7.1 Space + * TerraTec Aureon 7.1 Universe + * TerraTec Phase 22 + * TerraTec Phase 28 + * AudioTrak Prodigy 7.1 + * AudioTrak Prodigy 7.1 LT + * AudioTrak Prodigy 7.1 XT + * AudioTrak Prodigy 7.1 HIFI + * AudioTrak Prodigy 7.1 HD2 + * AudioTrak Prodigy 192 + * Pontis MS300 + * Albatron K8X800 Pro II + * Chaintech ZNF3-150 + * Chaintech ZNF3-250 + * Chaintech 9CJS + * Chaintech AV-710 + * Shuttle SN25P + * Onkyo SE-90PCI + * Onkyo SE-200PCI + * ESI Juli@ + * Hercules Fortissimo IV + * EGO-SYS WaveTerminal 192M + + model - Use the given board model, one of the following: + revo51, revo71, amp2000, prodigy71, prodigy71lt, + prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, + juli, aureon51, aureon71, universe, ap192, k8x800, + phase22, phase28, ms300, av710, se200pci, se90pci, + fortissimo4, sn25p, WT192M + + This module supports multiple cards and autoprobe. + + Note: The supported board is detected by reading EEPROM or PCI + SSID (if EEPROM isn't available). You can override the + model by passing "model" module option in case that the + driver isn't configured properly or you want to try another + type for testing. + + Module snd-indigo + ----------------- + + Module for Echoaudio Indigo + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-indigodj + ------------------- + + Module for Echoaudio Indigo DJ + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-indigoio + ------------------- + + Module for Echoaudio Indigo IO + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-intel8x0 + ------------------- + + Module for AC'97 motherboards from Intel and compatibles. + * Intel i810/810E, i815, i820, i830, i84x, MX440 + ICH5, ICH6, ICH7, 6300ESB, ESB2 + * SiS 7012 (SiS 735) + * NVidia NForce, NForce2, NForce3, MCP04, CK804 + CK8, CK8S, MCP501 + * AMD AMD768, AMD8111 + * ALi m5455 + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + buggy_irq - Enable workaround for buggy interrupts on some + motherboards (default yes on nForce chips, + otherwise off) + buggy_semaphore - Enable workaround for hardwares with buggy + semaphores (e.g. on some ASUS laptops) + (default off) + spdif_aclink - Use S/PDIF over AC-link instead of direct connection + from the controller chip + (0 = off, 1 = on, -1 = default) + + This module supports one chip and autoprobe. + + Note: the latest driver supports auto-detection of chip clock. + if you still encounter too fast playback, specify the clock + explicitly via the module option "ac97_clock=41194". + + Joystick/MIDI ports are not supported by this driver. If your + motherboard has these devices, use the ns558 or snd-mpu401 + modules, respectively. + + The power-management is supported. + + Module snd-intel8x0m + -------------------- + + Module for Intel ICH (i8x0) chipset MC97 modems. + * Intel i810/810E, i815, i820, i830, i84x, MX440 + ICH5, ICH6, ICH7 + * SiS 7013 (SiS 735) + * NVidia NForce, NForce2, NForce2s, NForce3 + * AMD AMD8111 + * ALi m5455 + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-interwave + -------------------- + + Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 + and other sound cards based on AMD InterWave (tm) chip. + + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) + pcm_voices - reserved PCM voices for the synthesizer (default 2) + effect - 1 = InterWave effects enable (default 0); + requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + + This module supports multiple cards, autoprobe and ISA PnP. + + Module snd-interwave-stb + ------------------------ + + Module for UltraSound 32-Pro (sound card from STB used by Compaq) + and other sound cards based on AMD InterWave (tm) chip with TEA6330T + circuit for extended control of bass, treble and master volume. + + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) + pcm_voices - reserved PCM voices for the synthesizer (default 2) + effect - 1 = InterWave effects enable (default 0); + requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + + This module supports multiple cards, autoprobe and ISA PnP. + + Module snd-korg1212 + ------------------- + + Module for Korg 1212 IO PCI card + + This module supports multiple cards. + + Module snd-layla20 + ------------------ + + Module for Echoaudio Layla20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-layla24 + ------------------ + + Module for Echoaudio Layla24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-maestro3 + ------------------- + + Module for Allegro/Maestro3 chips + + external_amp - enable external amp (enabled by default) + amp_gpio - GPIO pin number for external amp (0-15) or + -1 for default pin (8 for allegro, 1 for + others) + + This module supports autoprobe and multiple chips. + + Note: the binding of amplifier is dependent on hardware. + If there is no sound even though all channels are unmuted, try to + specify other gpio connection via amp_gpio option. + For example, a Panasonic notebook might need "amp_gpio=0x0d" + option. + + The power-management is supported. + + Module snd-mia + --------------- + + Module for Echoaudio Mia + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-miro + --------------- + + Module for Miro soundcards: miroSOUND PCM 1 pro, + miroSOUND PCM 12, + miroSOUND PCM 20 Radio. + + port - Port # (0x530,0x604,0xe80,0xf40) + irq - IRQ # (5,7,9,10,11) + dma1 - 1st dma # (0,1,3) + dma2 - 2nd dma # (0,1) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (5,7,9,10) + fm_port - FM Port # (0x388) + wss - enable WSS mode + ide - enable onboard ide support + + Module snd-mixart + ----------------- + + Module for Digigram miXart8 sound cards. + + This module supports multiple cards. + Note: One miXart8 board will be represented as 4 alsa cards. + See MIXART.txt for details. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via mixartloader utility in alsa-tools package. + + Module snd-mona + --------------- + + Module for Echoaudio Mona + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-mpu401 + ----------------- + + Module for MPU-401 UART devices. + + port - port number or -1 (disable) + irq - IRQ number or -1 (disable) + pnp - PnP detection - 0 = disable, 1 = enable (default) + + This module supports multiple devices and PnP. + + Module snd-mtpav + ---------------- + + Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel + port). + + port - I/O port # for MTPAV (0x378,0x278, default=0x378) + irq - IRQ # for MTPAV (7,5, default=7) + hwports - number of supported hardware ports, default=8. + + Module supports only 1 card. This module has no enable option. + + Module snd-mts64 + ---------------- + + Module for Ego Systems (ESI) Miditerminal 4140 + + This module supports multiple devices. + Requires parport (CONFIG_PARPORT). + + Module snd-nm256 + ---------------- + + Module for NeoMagic NM256AV/ZX chips + + playback_bufsize - max playback frame size in kB (4-128kB) + capture_bufsize - max capture frame size in kB (4-128kB) + force_ac97 - 0 or 1 (disabled by default) + buffer_top - specify buffer top address + use_cache - 0 or 1 (disabled by default) + vaio_hack - alias buffer_top=0x25a800 + reset_workaround - enable AC97 RESET workaround for some laptops + reset_workaround2 - enable extended AC97 RESET workaround for some + other laptops + + This module supports one chip and autoprobe. + + The power-management is supported. + + Note: on some notebooks the buffer address cannot be detected + automatically, or causes hang-up during initialization. + In such a case, specify the buffer top address explicitly via + the buffer_top option. + For example, + Sony F250: buffer_top=0x25a800 + Sony F270: buffer_top=0x272800 + The driver supports only ac97 codec. It's possible to force + to initialize/use ac97 although it's not detected. In such a + case, use force_ac97=1 option - but *NO* guarantee whether it + works! + + Note: The NM256 chip can be linked internally with non-AC97 + codecs. This driver supports only the AC97 codec, and won't work + with machines with other (most likely CS423x or OPL3SAx) chips, + even though the device is detected in lspci. In such a case, try + other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP + but some doesn't have ISA PnP. You'll need to specify isapnp=0 + and proper hardware parameters in the case without ISA PnP. + + Note: some laptops need a workaround for AC97 RESET. For the + known hardware like Dell Latitude LS and Sony PCG-F305, this + workaround is enabled automatically. For other laptops with a + hard freeze, you can try reset_workaround=1 option. + + Note: Dell Latitude CSx laptops have another problem regarding + AC97 RESET. On these laptops, reset_workaround2 option is + turned on as default. This option is worth to try if the + previous reset_workaround option doesn't help. + + Note: This driver is really crappy. It's a porting from the + OSS driver, which is a result of black-magic reverse engineering. + The detection of codec will fail if the driver is loaded *after* + X-server as described above. You might be able to force to load + the module, but it may result in hang-up. Hence, make sure that + you load this module *before* X if you encounter this kind of + problem. + + Module snd-opl3sa2 + ------------------ + + Module for Yamaha OPL3-SA2/SA3 sound cards. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - control port # for OPL3-SA chip (0x370) + sb_port - SB port # for OPL3-SA chip (0x220,0x240) + wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) + midi_port - port # for MPU-401 UART (0x300,0x330), -1 = disable + fm_port - FM port # for OPL3-SA chip (0x388), -1 = disable + irq - IRQ # for OPL3-SA chip (5,7,9,10) + dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3) + dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable + + This module supports multiple cards and ISA PnP. It does not support + autoprobe (if ISA PnP is not used) thus all ports must be specified!!! + + The power-management is supported. + + Module snd-opti92x-ad1848 + ------------------------- + + Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. + Module works with OAK Mozart cards as well. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-opti92x-cs4231 + ------------------------- + + Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + dma2 - second DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-opti93x + ------------------ + + Module for sound cards based on OPTi 82c93x chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + dma2 - second DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-oxygen + ----------------- + + Module for sound cards based on the C-Media CMI8788 chip: + * Asound A-8788 + * AuzenTech X-Meridian + * Bgears b-Enspirer + * Club3D Theatron DTS + * HT-Omega Claro + * Razer Barracuda AC-1 + * Sondigo Inferno + + This module supports autoprobe and multiple cards. + + Module snd-pcsp + ----------------- + + Module for internal PC-Speaker. + + nforce_wa - enable NForce chipset workaround. Expect bad sound. + + This module supports system beeps, some kind of PCM playback and + even a few mixer controls. + + Module snd-pcxhr + ---------------- + + Module for Digigram PCXHR boards + + This module supports multiple cards. + + Module snd-portman2x4 + --------------------- + + Module for Midiman Portman 2x4 parallel port MIDI interface + + This module supports multiple cards. + + Module snd-powermac (on ppc only) + --------------------------------- + + Module for PowerMac, iMac and iBook on-board soundchips + + enable_beep - enable beep using PCM (enabled as default) + + Module supports autoprobe a chip. + + Note: the driver may have problems regarding endianess. + + The power-management is supported. + + Module snd-pxa2xx-ac97 (on arm only) + ------------------------------------ + + Module for AC97 driver for the Intel PXA2xx chip + + For ARM architecture only. + + The power-management is supported. + + Module snd-riptide + ------------------ + + Module for Conexant Riptide chip + + joystick_port - Joystick port # (default: 0x200) + mpu_port - MPU401 port # (default: 0x330) + opl3_port - OPL3 port # (default: 0x388) + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + You need to install the firmware file "riptide.hex" to the standard + firmware path (e.g. /lib/firmware). + + Module snd-rme32 + ---------------- + + Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, + Prodif96 and Prodif Gold) sound cards. + + This module supports multiple cards. + + Module snd-rme96 + ---------------- + + Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. + + This module supports multiple cards. + + Module snd-rme9652 + ------------------ + + Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. + + precise_ptr - Enable precise pointer (doesn't work reliably). + (default = 0) + + This module supports multiple cards. + + Note: snd-page-alloc module does the job which snd-hammerfall-mem + module did formerly. It will allocate the buffers in advance + when any RME9652 cards are found. To make the buffer + allocation sure, load snd-page-alloc module in the early + stage of boot sequence. See "Early Buffer Allocation" + section. + + Module snd-sa11xx-uda1341 (on arm only) + --------------------------------------- + + Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. + + Module supports only one card. + Module has no enable and index options. + + The power-management is supported. + + Module snd-sb8 + -------------- + + Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, + SoundBlaster 2.0, + SoundBlaster Pro + + port - port # for SB DSP chip (0x220,0x240,0x260) + irq - IRQ # for SB DSP chip (5,7,9,10) + dma8 - DMA # for SB DSP chip (1,3) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-sb16 and snd-sbawe + ----------------------------- + + Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), + SoundBlaster AWE 32 (PnP), + SoundBlaster AWE 64 PnP + + mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) + csp - ASP/CSP chip support - 0 = disable (default), 1 = enable + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for SB DSP 4.x chip (0x220,0x240,0x260) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable + awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) + (snd-sbawe module only) + irq - IRQ # for SB DSP 4.x chip (5,7,9,10) + dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) + dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) + + This module supports multiple cards, autoprobe and ISA PnP. + + Note: To use Vibra16X cards in 16-bit half duplex mode, you must + disable 16bit DMA with dma16 = -1 module parameter. + Also, all Sound Blaster 16 type cards can operate in 16-bit + half duplex mode through 8-bit DMA channel by disabling their + 16-bit DMA channel. + + The power-management is supported. + + Module snd-sc6000 + ----------------- + + Module for Gallant SC-6000 soundcard. + + port - Port # (0x220 or 0x240) + mss_port - MSS Port # (0x530 or 0xe80) + irq - IRQ # (5,7,9,10,11) + mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq + dma - DMA # (1,3,0) + + This module supports multiple cards. + + This card is also known as Audio Excel DSP 16 or Zoltrix AV302. + + Module snd-sgalaxy + ------------------ + + Module for Aztech Sound Galaxy sound card. + + sbport - Port # for SB16 interface (0x220,0x240) + wssport - Port # for WSS interface (0x530,0xe80,0xf40,0x604) + irq - IRQ # (7,9,10,11) + dma1 - DMA # + + This module supports multiple cards. + + The power-management is supported. + + Module snd-sscape + ----------------- + + Module for ENSONIQ SoundScape PnP cards. + + port - Port # (PnP setup) + wss_port - WSS Port # (PnP setup) + irq - IRQ # (PnP setup) + mpu_irq - MPU-401 IRQ # (PnP setup) + dma - DMA # (PnP setup) + dma2 - 2nd DMA # (PnP setup, -1 to disable) + + This module supports multiple cards. ISA PnP must be enabled. + You need sscape_ctl tool in alsa-tools package for loading + the microcode. + + Module snd-sun-amd7930 (on sparc only) + -------------------------------------- + + Module for AMD7930 sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-sun-cs4231 (on sparc only) + ------------------------------------- + + Module for CS4231 sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-sun-dbri (on sparc only) + ----------------------------------- + + Module for DBRI sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-wavefront + -------------------- + + Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + + use_cs4232_midi - Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + cs4232_pcm_port - Port # for CS4232 PCM interface. + cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). + cs4232_mpu_port - Port # for CS4232 MPU-401 interface. + cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). + ics2115_port - Port # for ICS2115 + ics2115_irq - IRQ # for ICS2115 + fm_port - FM OPL-3 Port # + dma1 - DMA1 # for CS4232 PCM interface. + dma2 - DMA2 # for CS4232 PCM interface. + + The below are options for wavefront_synth features: + wf_raw - Assume that we need to boot the OS (default:no) + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. + fx_raw - Assume that the FX process needs help (default:yes) + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. + debug_default - Debug parameters for card initialization + wait_usecs - How long to wait without sleeping, usecs + (default:150) + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. + sleep_interval - How long to sleep when waiting for reply + (default: 100) + sleep_tries - How many times to try sleeping during a wait + (default: 50) + ospath - Pathname to processed ICS2115 OS firmware + (default:wavefront.os) + The path name of the ISC2115 OS firmware. In the recent + version, it's handled via firmware loader framework, so it + must be installed in the proper path, typically, + /lib/firmware. + reset_time - How long to wait for a reset to take effect + (default:2) + ramcheck_time - How many seconds to wait for the RAM test + (default:20) + osrun_time - How many seconds to wait for the ICS2115 OS + (default:10) + + This module supports multiple cards and ISA PnP. + + Note: the firmware file "wavefront.os" was located in the earlier + version in /etc. Now it's loaded via firmware loader, and + must be in the proper firmware path, such as /lib/firmware. + Copy (or symlink) the file appropriately if you get an error + regarding firmware downloading after upgrading the kernel. + + Module snd-sonicvibes + --------------------- + + Module for S3 SonicVibes PCI sound cards. + * PINE Schubert 32 PCI + + reverb - Reverb Enable - 1 = enable, 0 = disable (default) + - SoundCard must have onboard SRAM for this. + mge - Mic Gain Enable - 1 = enable, 0 = disable (default) + + This module supports multiple cards and autoprobe. + + Module snd-serial-u16550 + ------------------------ + + Module for UART16550A serial MIDI ports. + + port - port # for UART16550A chip + irq - IRQ # for UART16550A chip, -1 = poll mode + speed - speed in bauds (9600,19200,38400,57600,115200) + 38400 = default + base - base for divisor in bauds (57600,115200,230400,460800) + 115200 = default + outs - number of MIDI ports in a serial port (1-4) + 1 = default + adaptor - Type of adaptor. + 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, + 3 = MS-124W M/B, 4 = Generic + + This module supports multiple cards. This module does not support autoprobe + thus the main port must be specified!!! Other options are optional. + + Module snd-trident + ------------------ + + Module for Trident 4DWave DX/NX sound cards. + * Best Union Miss Melody 4DWave PCI + * HIS 4DWave PCI + * Warpspeed ONSpeed 4DWave PCI + * AzTech PCI 64-Q3D + * Addonics SV 750 + * CHIC True Sound 4Dwave + * Shark Predator4D-PCI + * Jaton SonicWave 4D + * SiS SI7018 PCI Audio + * Hoontech SoundTrack Digital 4DWave NX + + pcm_channels - max channels (voices) reserved for PCM + wavetable_size - max wavetable size in kB (4-?kb) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-usb-audio + -------------------- + + Module for USB audio and USB MIDI devices. + + vid - Vendor ID for the device (optional) + pid - Product ID for the device (optional) + nrpacks - Max. number of packets per URB (default: 8) + async_unlink - Use async unlink mode (default: yes) + device_setup - Device specific magic number (optional) + - Influence depends on the device + - Default: 0x0000 + ignore_ctl_error - Ignore any USB-controller regarding mixer + interface (default: no) + + This module supports multiple devices, autoprobe and hotplugging. + + NB: nrpacks parameter can be modified dynamically via sysfs. + Don't put the value over 20. Changing via sysfs has no sanity + check. + NB: async_unlink=0 would cause Oops. It remains just for + debugging purpose (if any). + NB: ignore_ctl_error=1 may help when you get an error at accessing + the mixer element such as URB error -22. This happens on some + buggy USB device or the controller. + + Module snd-usb-caiaq + -------------------- + + Module for caiaq UB audio interfaces, + * Native Instruments RigKontrol2 + * Native Instruments Kore Controller + * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 8 DJ + + This module supports multiple devices, autoprobe and hotplugging. + + Module snd-usb-usx2y + -------------------- + + Module for Tascam USB US-122, US-224 and US-428 devices. + + This module supports multiple devices, autoprobe and hotplugging. + + Note: you need to load the firmware via usx2yloader utility included + in alsa-tools and alsa-firmware packages. + + Module snd-via82xx + ------------------ + + Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, + 8233A, 8233C, 8235, 8237 (south) bridge. + + mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup + [VIA686A/686B only] + joystick - Enable joystick (default off) [VIA686A/686B only] + ac97_clock - AC'97 codec clock base (default 48000Hz) + dxs_support - support DXS channels, + 0 = auto (default), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA, 5 = enable any sample + rate and different sample rates on different + channels + [VIA8233/C, 8235, 8237 only] + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + + This module supports one chip and autoprobe. + + Note: on some SMP motherboards like MSI 694D the interrupts might + not be generated properly. In such a case, please try to + set the SMP (or MPS) version on BIOS to 1.1 instead of + default value 1.4. Then the interrupt number will be + assigned under 15. You might also upgrade your BIOS. + + Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) + channels as the first PCM. On these channels, up to 4 + streams can be played at the same time, and the controller + can perform sample rate conversion with separate rates for + each channel. + As default (dxs_support = 0), 48k fixed rate is chosen + except for the known devices since the output is often + noisy except for 48k on some mother boards due to the + bug of BIOS. + Please try once dxs_support=5 and if it works on other + sample rates (e.g. 44.1kHz of mp3 playback), please let us + know the PCI subsystem vendor/device id's (output of + "lspci -nv"). + If dxs_support=5 does not work, try dxs_support=4; if it + doesn't work too, try dxs_support=1. (dxs_support=1 is + usually for old motherboards. The correct implemented + board should work with 4 or 5.) If it still doesn't + work and the default setting is ok, dxs_support=3 is the + right choice. If the default setting doesn't work at all, + try dxs_support=2 to disable the DXS channels. + In any cases, please let us know the result and the + subsystem vendor/device ids. See "Links and Addresses" + below. + + Note: for the MPU401 on VIA823x, use snd-mpu401 driver + additionally. The mpu_port option is for VIA686 chips only. + + The power-management is supported. + + Module snd-via82xx-modem + ------------------------ + + Module for VIA82xx AC97 modem + + ac97_clock - AC'97 codec clock base (default 48000Hz) + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-virmidi + ------------------ + + Module for virtual rawmidi devices. + This module creates virtual rawmidi devices which communicate + to the corresponding ALSA sequencer ports. + + midi_devs - MIDI devices # (1-4, default=4) + + This module supports multiple cards. + + Module snd-virtuoso + ------------------- + + Module for sound cards based on the Asus AV100/AV200 chips, + i.e., Xonar D1, DX, D2, D2X and HDAV1.3 (Deluxe). + + This module supports autoprobe and multiple cards. + + Module snd-vx222 + ---------------- + + Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. + + mic - Enable Microphone on V222 Mic (NYI) + ibl - Capture IBL size. (default = 0, minimum size) + + This module supports multiple cards. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via vxloader utility in alsa-tools package. To invoke + vxloader automatically, add the following to /etc/modprobe.conf + + install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader + + (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to + /etc/modules.conf, instead.) + IBL size defines the interrupts period for PCM. The smaller size + gives smaller latency but leads to more CPU consumption, too. + The size is usually aligned to 126. As default (=0), the smallest + size is chosen. The possible IBL values can be found in + /proc/asound/cardX/vx-status proc file. + + The power-management is supported. + + Module snd-vxpocket + ------------------- + + Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. + + ibl - Capture IBL size. (default = 0, minimum size) + + This module supports multiple cards. The module is compiled only when + PCMCIA is supported on kernel. + + With the older 2.6.x kernel, to activate the driver via the card + manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the + sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no + longer require a config file. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via vxloader utility in alsa-tools package. + + About capture IBL, see the description of snd-vx222 module. + + Note: snd-vxp440 driver is merged to snd-vxpocket driver since + ALSA 1.0.10. + + The power-management is supported. + + Module snd-ymfpci + ----------------- + + Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). + + mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default, + 1 (auto-detect for YMF744/754 only) + fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default + 1 (auto-detect for YMF744/754 only) + joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default, + 1 (auto-detect) + rear_switch - enable shared rear/line-in switch (bool) + + This module supports autoprobe and multiple chips. + + The power-management is supported. + + Module snd-pdaudiocf + -------------------- + + Module for Sound Core PDAudioCF sound card. + + The power-management is supported. + + +AC97 Quirk Option +================= + +The ac97_quirk option is used to enable/override the workaround for +specific devices on drivers for on-board AC'97 controllers like +snd-intel8x0. Some hardware have swapped output pins between Master +and Headphone, or Surround (thanks to confusion of AC'97 +specifications from version to version :-) + +The driver provides the auto-detection of known problematic devices, +but some might be unknown or wrongly detected. In such a case, pass +the proper value with this option. + +The following strings are accepted: + - default Don't override the default setting + - none Disable the quirk + - hp_only Bind Master and Headphone controls as a single control + - swap_hp Swap headphone and master controls + - swap_surround Swap master and surround controls + - ad_sharing For AD1985, turn on OMS bit and use headphone + - alc_jack For ALC65x, turn on the jack sense mode + - inv_eapd Inverted EAPD implementation + - mute_led Bind EAPD bit for turning on/off mute LED + +For backward compatibility, the corresponding integer value -1, 0, +... are accepted, too. + +For example, if "Master" volume control has no effect on your device +but only "Headphone" does, pass ac97_quirk=hp_only module option. + + +Configuring Non-ISAPNP Cards +============================ + +When the kernel is configured with ISA-PnP support, the modules +supporting the isapnp cards will have module options "isapnp". +If this option is set, *only* the ISA-PnP devices will be probed. +For probing the non ISA-PnP cards, you have to pass "isapnp=0" option +together with the proper i/o and irq configuration. + +When the kernel is configured without ISA-PnP support, isapnp option +will be not built in. + + +Module Autoloading Support +========================== + +The ALSA drivers can be loaded automatically on demand by defining +module aliases. The string 'snd-card-%1' is requested for ALSA native +devices where %i is sound card number from zero to seven. + +To auto-load an ALSA driver for OSS services, define the string +'sound-slot-%i' where %i means the slot number for OSS, which +corresponds to the card index of ALSA. Usually, define this +as the same card module. + +An example configuration for a single emu10k1 card is like below: +----- /etc/modprobe.conf +alias snd-card-0 snd-emu10k1 +alias sound-slot-0 snd-emu10k1 +----- /etc/modprobe.conf + +The available number of auto-loaded sound cards depends on the module +option "cards_limit" of snd module. As default it's set to 1. +To enable the auto-loading of multiple cards, specify the number of +sound cards in that option. + +When multiple cards are available, it'd better to specify the index +number for each card via module option, too, so that the order of +cards is kept consistent. + +An example configuration for two sound cards is like below: + +----- /etc/modprobe.conf +# ALSA portion +options snd cards_limit=2 +alias snd-card-0 snd-interwave +alias snd-card-1 snd-ens1371 +options snd-interwave index=0 +options snd-ens1371 index=1 +# OSS/Free portion +alias sound-slot-0 snd-interwave +alias sound-slot-1 snd-ens1371 +----- /etc/modprobe.conf + +In this example, the interwave card is always loaded as the first card +(index 0) and ens1371 as the second (index 1). + +Alternative (and new) way to fixate the slot assignment is to use +"slots" option of snd module. In the case above, specify like the +following: + +options snd slots=snd-interwave,snd-ens1371 + +Then, the first slot (#0) is reserved for snd-interwave driver, and +the second (#1) for snd-ens1371. You can omit index option in each +driver if slots option is used (although you can still have them at +the same time as long as they don't conflict). + +The slots option is especially useful for avoiding the possible +hot-plugging and the resultant slot conflict. For example, in the +case above again, the first two slots are already reserved. If any +other driver (e.g. snd-usb-audio) is loaded before snd-interwave or +snd-ens1371, it will be assigned to the third or later slot. + +When a module name is given with '!', the slot will be given for any +modules but that name. For example, "slots=!snd-pcsp" will reserve +the first slot for any modules but snd-pcsp. + + +ALSA PCM devices to OSS devices mapping +======================================= + +/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 +/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 +/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 +/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 +/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 +/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 +/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 +/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 +/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 + +The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means +sound card number and second means device number. The ALSA devices +have either 'c' or 'p' suffix indicating the direction, capture and +playback, respectively. + +Please note that the device mapping above may be varied via the module +options of snd-pcm-oss module. + + +Proc interfaces (/proc/asound) +============================== + +/proc/asound/card#/pcm#[cp]/oss +------------------------------- + String "erase" - erase all additional informations about OSS applications + String "<app_name> <fragments> <fragment_size> [<options>]" + + <app_name> - name of application with (higher priority) or without path + <fragments> - number of fragments or zero if auto + <fragment_size> - size of fragment in bytes or zero if auto + <options> - optional parameters + - disable the application tries to open a pcm device for + this channel but does not want to use it. + (Cause a bug or mmap needs) + It's good for Quake etc... + - direct don't use plugins + - block force block mode (rvplayer) + - non-block force non-block mode + - whole-frag write only whole fragments (optimization affecting + playback only) + - no-silence do not fill silence ahead to avoid clicks + - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl + instead of filled blocks + + Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss + echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss + echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss + + +Early Buffer Allocation +======================= + +Some drivers (e.g. hdsp) require the large contiguous buffers, and +sometimes it's too late to find such spaces when the driver module is +actually loaded due to memory fragmentation. You can pre-allocate the +PCM buffers by loading snd-page-alloc module and write commands to its +proc file in prior, for example, in the early boot stage like +/etc/init.d/*.local scripts. + +Reading the proc file /proc/drivers/snd-page-alloc shows the current +usage of page allocation. In writing, you can send the following +commands to the snd-page-alloc driver: + + - add VENDOR DEVICE MASK SIZE BUFFERS + + VENDOR and DEVICE are PCI vendor and device IDs. They take + integer numbers (0x prefix is needed for the hex). + MASK is the PCI DMA mask. Pass 0 if not restricted. + SIZE is the size of each buffer to allocate. You can pass + k and m suffix for KB and MB. The max number is 16MB. + BUFFERS is the number of buffers to allocate. It must be greater + than 0. The max number is 4. + + - erase + + This will erase the all pre-allocated buffers which are not in + use. + + +Links and Addresses +=================== + + ALSA project homepage + http://www.alsa-project.org + + ALSA Bug Tracking System + https://bugtrack.alsa-project.org/bugs/ + + ALSA Developers ML + mailto:alsa-devel@alsa-project.org diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt new file mode 100644 index 0000000..7f10dc6 --- /dev/null +++ b/Documentation/sound/alsa/Audigy-mixer.txt @@ -0,0 +1,345 @@ + + Sound Blaster Audigy mixer / default DSP code + =========================================== + +This is based on SB-Live-mixer.txt. + +The EMU10K2 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K2 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + +name='PCM Front Playback Volume',index=0 + +This control is used to attenuate samples for left and right front PCM FX-bus +accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM +samples for 5.1 playback. The result samples are forwarded to the front DAC PCM +slots of the Philips DAC. + +name='PCM Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right surround PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM +samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM +slots of the Philips DAC. + +name='PCM Center Playback Volume',index=0 + +This control is used to attenuate samples for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample +is forwarded to the center DAC PCM slot of the Philips DAC. + +name='PCM LFE Playback Volume',index=0 + +This control is used to attenuate sample for LFE PCM FX-bus accumulator. +ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample +is forwarded to the LFE DAC PCM slot of the Philips DAC. + +name='PCM Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for +stereo playback. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='PCM Capture Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Music Playback Volume',index=0 + +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Music Capture Volume',index=0 + +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Mic Playback Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +For Mic input is used AC97 codec. The result samples are forwarded to +the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic +capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. + +name='Mic Capture Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Audigy CD Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the Philips DAC. + +name='Audigy CD Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 Optical Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='IEC958 Optical Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Line2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Analog Mix Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs from Philips ADC. The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. This contains mix from analog sources +like CD, Line In, Aux, .... + +name='Analog Mix Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs Philips ADC. The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Aux2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Aux2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Front Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right front speakers of +this mix. + +name='Surround Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right surround speakers of +this mix. + +name='Center Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for center speaker of this mix. + +name='LFE Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for LFE speaker of this mix. + +name='Tone Control - Switch',index=0 + +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 + +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 + +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Master Playback Volume',index=0 + +This control is used to attenuate samples for front, surround, center and +LFE outputs. + +name='IEC958 Optical Raw Playback Switch',index=0 + +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + + +2) PCM stream related controls +------------------------------ + +name='EMU10K1 PCM Volume',index 0-31 + +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + + 0 - mono, default 0xffff (no attenuation) + 1 - left, default 0xffff (no attenuation) + 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 + +This control specifies the destination - FX-bus accumulators. There 24 +values with this mapping: + + 0 - mono, A destination (FX-bus 0-63), default 0 + 1 - mono, B destination (FX-bus 0-63), default 1 + 2 - mono, C destination (FX-bus 0-63), default 2 + 3 - mono, D destination (FX-bus 0-63), default 3 + 4 - mono, E destination (FX-bus 0-63), default 0 + 5 - mono, F destination (FX-bus 0-63), default 0 + 6 - mono, G destination (FX-bus 0-63), default 0 + 7 - mono, H destination (FX-bus 0-63), default 0 + 8 - left, A destination (FX-bus 0-63), default 0 + 9 - left, B destination (FX-bus 0-63), default 1 + 10 - left, C destination (FX-bus 0-63), default 2 + 11 - left, D destination (FX-bus 0-63), default 3 + 12 - left, E destination (FX-bus 0-63), default 0 + 13 - left, F destination (FX-bus 0-63), default 0 + 14 - left, G destination (FX-bus 0-63), default 0 + 15 - left, H destination (FX-bus 0-63), default 0 + 16 - right, A destination (FX-bus 0-63), default 0 + 17 - right, B destination (FX-bus 0-63), default 1 + 18 - right, C destination (FX-bus 0-63), default 2 + 19 - right, D destination (FX-bus 0-63), default 3 + 20 - right, E destination (FX-bus 0-63), default 0 + 21 - right, F destination (FX-bus 0-63), default 0 + 22 - right, G destination (FX-bus 0-63), default 0 + 23 - right, H destination (FX-bus 0-63), default 0 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 + +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + + 0 - mono, A destination attn, default 255 (no attenuation) + 1 - mono, B destination attn, default 255 (no attenuation) + 2 - mono, C destination attn, default 0 (mute) + 3 - mono, D destination attn, default 0 (mute) + 4 - mono, E destination attn, default 0 (mute) + 5 - mono, F destination attn, default 0 (mute) + 6 - mono, G destination attn, default 0 (mute) + 7 - mono, H destination attn, default 0 (mute) + 8 - left, A destination attn, default 255 (no attenuation) + 9 - left, B destination attn, default 0 (mute) + 10 - left, C destination attn, default 0 (mute) + 11 - left, D destination attn, default 0 (mute) + 12 - left, E destination attn, default 0 (mute) + 13 - left, F destination attn, default 0 (mute) + 14 - left, G destination attn, default 0 (mute) + 15 - left, H destination attn, default 0 (mute) + 16 - right, A destination attn, default 0 (mute) + 17 - right, B destination attn, default 255 (no attenuation) + 18 - right, C destination attn, default 0 (mute) + 19 - right, D destination attn, default 0 (mute) + 20 - right, E destination attn, default 0 (mute) + 21 - right, F destination attn, default 0 (mute) + 22 - right, G destination attn, default 0 (mute) + 23 - right, H destination attn, default 0 (mute) + + + +4) MANUALS/PATENTS: +------------------- + +ftp://opensource.creative.com/pub/doc +------------------------------------- + + Files: + LM4545.pdf AC97 Codec + + m2049.pdf The EMU10K1 Digital Audio Processor + + hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + Patent numbers: + WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) + streams + + WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + + WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + + US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + + US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + + US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + + US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) + + US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + + US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) + pool of short term memory registers + + US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 0000000..a4c53d8 --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,442 @@ + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 + ======================================================== + + Thibault Le Meur <Thibault.LeMeur@supelec.fr> + +This document is a guide to using the M-Audio Audiophile USB (tm) device with +ALSA and JACK. + +History +======= +* v1.4 - Thibault Le Meur (2007-07-11) + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> + - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + +1 - Audiophile USB Specs and correct usage +========================================== + +This part is a reminder of important facts about the functions and limitations +of the device. + +The device has 4 audio interfaces, and 2 MIDI ports: + * Analog Stereo Input (Ai) + - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) + - When the 1/4" TS (jack) connectors are connected, the RCA connectors + are disabled + * Analog Stereo Output (Ao) + * Digital Stereo Input (Di) + * Digital Stereo Output (Do) + * Midi In (Mi) + * Midi Out (Mo) + +The internal DAC/ADC has the following characteristics: +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" + +Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be +activated at the same time depending on the audio mode selected: + * 16-bit/48kHz ==> 4 channels in + 4 channels out + - Ai+Ao+Di+Do + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out + - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) + - Ai or Ao or Di or Do + +Important facts about the Digital interface: +-------------------------------------------- + * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under Linux + - Note that in this setup only the Do interface can be enabled + * Apart from recording an audio digital stream, enabling the Di port is a way +to synchronize the device to an external sample clock + - As a consequence, the Di port must be enable only if an active Digital +source is connected + - Enabling Di when no digital source is connected can result in a +synchronization error (for instance sound played at an odd sample rate) + + +2 - Audiophile USB MIDI support in ALSA +======================================= + +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + * snd-usb-audio + * snd-seq-midi + +No additional setting is required. + + +3 - Audiophile USB Audio support in ALSA +======================================== + +Audio functions of the Audiophile USB device are handled by the snd-usb-audio +module. This module can work in a default mode (without any device-specific +parameter), or in an "advanced" mode with the device-specific parameter called +"device_setup". + +3.1 - Default Alsa driver mode +------------------------------ + +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): + * hw:1,0 is Ao in playback and Di in capture + * hw:1,1 is Do in playback and Ai in capture + * hw:1,2 is Do in AC3/DTS passthrough mode + +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. + +Examples: + * playing a S24_3BE encoded raw file to the Ao port + % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw + * recording a S24_3BE encoded raw file from the Ai port + % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw + * playing a S16_BE encoded raw file to the Do port + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw + +If you're happy with the default Alsa driver mode and don't experience any +issue with this mode, then you can skip the following chapter. + +3.2 - Advanced module setup +--------------------------- + +Due to the hardware constraints described above, the device initialization made +by the Alsa driver in default mode may result in a corrupted state of the +device. For instance, a particularly annoying issue is that the sound captured +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). + +For people having this problem, the snd-usb-audio module has a new module +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) + +3.2.1 - Initializing the working mode of the Audiophile USB + +As far as the Audiophile USB device is concerned, this value let the user +specify: + * the sample depth + * the sample rate + * whether the Di port is used or not + +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + + * device_setup=0x01 + - 16bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x11 + - 16bits 48kHz mode with Di enabled + - Ai,Ao,Di,Do can be used at the same time + - hw:1,0 is available in capture mode + - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + + * device_setup=0x09 + - 24bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x19 + - 24bits 48kHz mode with Di enabled + - 3 ports from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in capture mode and an active digital source must be + connected to Di + - hw:1,2 is not available + + * device_setup=0x0D or 0x10 + - 24bits 96kHz mode + - Di is enabled by default for this mode but does not need to be connected + to an active source + - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in captured mode + - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + + * device_setup=0x03 + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +3.2.2 - How to use the device_setup parameter +---------------------------------------------- + +The parameter can be given: + + * By manually probing the device (as root): + # modprobe -r snd-usb-audio + # modprobe snd-usb-audio index=1 device_setup=0x09 + + * Or while configuring the modules options in your modules configuration file + - For Fedora distributions, edit the /etc/modprobe.conf file: + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +CAUTION when initializing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead to a misconfiguration of the device. In this case + turn off the device, unprobe the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: + - first turn off the device + - de-register the snd-usb-audio module (modprobe -r) + - change the device_setup parameter by changing the device_setup + option in /etc/modprobe.conf + - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. + +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure + +If you want to understand the device_setup magic numbers for the Audiophile +USB, you need some very basic understanding of binary computation. However, +this is not required to use the parameter and you may skip this section. + +The device_setup is one byte long and its structure is the following: + + +---+---+---+---+---+---+---+---+ + | b7| b6| b5| b4| b3| b2| b1| b0| + +---+---+---+---+---+---+---+---+ + | 0 | 0 | 0 | Di|24B|96K|DTS|SET| + +---+---+---+---+---+---+---+---+ + +Where: + * b0 is the "SET" bit + - it MUST be set if device_setup is initialized + * b1 is the "DTS" bit + - it is set only for Digital output with DTS/AC3 + - this setup is not tested + * b2 is the Rate selection flag + - When set to "1" the rate range is 48.1-96kHz + - Otherwise the sample rate range is 8-48kHz + * b3 is the bit depth selection flag + - When set to "1" samples are 24bits long + - Otherwise they are 16bits long + - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits + samples + * b4 is the Digital input flag + - When set to "1" the device assumes that an active digital source is + connected + - You shouldn't enable Di if no source is seen on the port (this leads to + synchronization issues) + - b4 is implied by b2 (since only one port is enabled at a time no synch + error can occur) + * b5 to b7 are reserved for future uses, and must be set to "0" + - might become Ao, Do, Ai, for b7, b6, b4 respectively + +Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since + b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll + only be able to use one at the same time + +3.2.3.2 - USB implementation details for this device + +You may safely skip this section if you're not interested in driver +hacking. + +This section describes some internal aspects of the device and summarizes the +data I got by usb-snooping the windows and Linux drivers. + +The M-Audio Audiophile USB has 7 USB Interfaces: +a "USB interface": + * USB Interface nb.0 + * USB Interface nb.1 + - Audio Control function + * USB Interface nb.2 + - Analog Output + * USB Interface nb.3 + - Digital Output + * USB Interface nb.4 + - Analog Input + * USB Interface nb.5 + - Digital Input + * USB Interface nb.6 + - MIDI interface compliant with the MIDIMAN quirk + +Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: + * Interface 3 (Digital Out) has an extra Alset nb.6 + * Interface 5 (Digital In) does not have Alset nb.3 and 5 + +Here is a short description of the AltSettings capabilities: + * AltSettings 1 corresponds to + - 24-bit depth, 48.1-96kHz sample mode + - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) + * AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 6 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch playback (Do), audio format type III IEC1937_AC-3 + +In order to ensure a correct initialization of the device, the driver +_must_know_ how the device will be used: + * if DTS is chosen, only Interface 2 with AltSet nb.6 must be + registered + * if 96KHz only AltSets nb.1 of each interface must be selected + * if samples are using 24bits/48KHz then AltSet 2 must me used if + Digital input is connected, and only AltSet nb.3 if Digital input + is not connected + * if samples are using 16bits/48KHz then AltSet 4 must me used if + Digital input is connected, and only AltSet nb.5 if Digital input + is not connected + +When device_setup is given as a parameter to the snd-usb-audio module, the +parse_audio_endpoints function uses a quirk called +"audiophile_skip_setting_quirk" in order to prevent AltSettings not +corresponding to device_setup from being registered in the driver. + +4 - Audiophile USB and Jack support +=================================== + +This section deals with support of the Audiophile USB device in Jack. + +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +4.1 - Direct support in Jackd +----------------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +exactly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. + +For instance here is one way to run Jack with 2 playback channels on Ao and 2 +capture channels from Ai: + % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 + +However you may see the following warning message: +"You appear to be using the ALSA software "plug" layer, probably a result of +using the "default" ALSA device. This is less efficient than it could be. +Consider using a hardware device instead rather than using the plug layer." + +4.3 - Getting 2 input and/or output interfaces in Jack +------------------------------------------------------ + +As you can see, starting the Jack server this way will only enable 1 stereo +input (Di or Ai) and 1 stereo output (Ao or Do). + +This is due to the following restrictions: +* Jack can only open one capture device and one playback device at a time +* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 + (and optionally hw:1,2) + +If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to +combine the Alsa devices into one logical "complex" device. + +If you want to give it a try, I recommend reading the information from +this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html +It is related to another device (ice1712) but can be adapted to suit +the Audiophile USB. + +Enabling multiple Audiophile USB interfaces for Jackd will certainly require: +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc + file +* start jackd with this device + +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt new file mode 100644 index 0000000..f158cde --- /dev/null +++ b/Documentation/sound/alsa/Bt87x.txt @@ -0,0 +1,78 @@ +Intro +===== + +You might have noticed that the bt878 grabber cards have actually +_two_ PCI functions: + +$ lspci +[ ... ] +00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) +00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02) +[ ... ] + +The first does video, it is backward compatible to the bt848. The second +does audio. snd-bt87x is a driver for the second function. It's a sound +driver which can be used for recording sound (and _only_ recording, no +playback). As most TV cards come with a short cable which can be plugged +into your sound card's line-in you probably don't need this driver if all +you want to do is just watching TV... + +Some cards do not bother to connect anything to the audio input pins of +the chip, and some other cards use the audio function to transport MPEG +video data, so it's quite possible that audio recording may not work +with your card. + + +Driver Status +============= + +The driver is now stable. However, it doesn't know about many TV cards, +and it refuses to load for cards it doesn't know. + +If the driver complains ("Unknown TV card found, the audio driver will +not load"), you can specify the load_all=1 option to force the driver to +try to use the audio capture function of your card. If the frequency of +recorded data is not right, try to specify the digital_rate option with +other values than the default 32000 (often it's 44100 or 64000). + +If you have an unknown card, please mail the ID and board name to +<alsa-devel@alsa-project.org>, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. + + +Audio modes +=========== + +The chip knows two different modes (digital/analog). snd-bt87x +registers two PCM devices, one for each mode. They cannot be used at +the same time. + + +Digital audio mode +================== + +The first device (hw:X,0) gives you 16 bit stereo sound. The sample +rate depends on the external source which feeds the Bt87x with digital +sound via I2S interface. + + +Analog audio mode (A/D) +======================= + +The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported +sample rates are between 119466 and 448000 Hz (yes, these numbers are +that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the +maximum sample rate is 1792000 Hz, but audio data becomes unusable +beyond 896000 Hz on my card. + +The chip has three analog inputs. Consequently you'll get a mixer +device to control these. + + +Have fun, + + Clemens + + +Written by Clemens Ladisch <clemens@ladisch.de> +big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org> diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt new file mode 100644 index 0000000..16935c8 --- /dev/null +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -0,0 +1,254 @@ + Brief Notes on C-Media 8338/8738/8768/8770 Driver + ================================================= + + Takashi Iwai <tiwai@suse.de> + + +Front/Rear Multi-channel Playback +--------------------------------- + +CM8x38 chip can use ADC as the second DAC so that two different stereo +channels can be used for front/rear playbacks. Since there are two +DACs, both streams are handled independently unlike the 4/6ch multi- +channel playbacks in the section below. + +As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for +card#0) for front and 4/6ch playbacks, while the second PCM device +(hw:0,1) is assigned to the second DAC for rear playback. + +There are slight differences between the two DACs: + +- The first DAC supports U8 and S16LE formats, while the second DAC + supports only S16LE. +- The second DAC supports only two channel stereo. + +Please note that the CM8x38 DAC doesn't support continuous playback +rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000, +44100 and 48000 Hz. + +The rear output can be heard only when "Four Channel Mode" switch is +disabled. Otherwise no signal will be routed to the rear speakers. +As default it's turned on. + +*** WARNING *** +When "Four Channel Mode" switch is off, the output from rear speakers +will be FULL VOLUME regardless of Master and PCM volumes. +This might damage your audio equipment. Please disconnect speakers +before your turn off this switch. +*** WARNING *** + +[ Well.. I once got the output with correct volume (i.e. same with the + front one) and was so excited. It was even with "Four Channel" bit + on and "double DAC" mode. Actually I could hear separate 4 channels + from front and rear speakers! But.. after reboot, all was gone. + It's a very pity that I didn't save the register dump at that + time.. Maybe there is an unknown register to achieve this... ] + +If your card has an extra output jack for the rear output, the rear +playback should be routed there as default. If not, there is a +control switch in the driver "Line-In As Rear", which you can change +via alsamixer or somewhat else. When this switch is on, line-in jack +is used as rear output. + +There are two more controls regarding to the rear output. +The "Exchange DAC" switch is used to exchange front and rear playback +routes, i.e. the 2nd DAC is output from front output. + + +4/6 Multi-Channel Playback +-------------------------- + +The recent CM8738 chips support for the 4/6 multi-channel playback +function. This is useful especially for AC3 decoding. + +When the multi-channel is supported, the driver name has a suffix +"-MC" such like "CMI8738-MC6". You can check this name from +/proc/asound/cards. + +When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or +4) channels. While the dual DAC supports two different rates or +formats, the 4/6-ch playback supports only the same condition for all +channels. Since the multi-channel playback mode uses both DACs, you +cannot operate with full-duplex. + +The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" +in alsa-lib. For example, you can play a WAV file with 6 channels like + + % aplay -Dsurround51 sixchannels.wav + +For programming the 4/6 channel playback, you need to specify the PCM +channels as you like and set the format S16LE. For example, for playback +with 4 channels, + + snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); + // or mmap if you like + snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE); + snd_pcm_hw_params_set_channels(pcm, hw, 4); + +and use the interleaved 4 channel data. + +There are some control switchs affecting to the speaker connections: + +"Line-In Mode" - an enum control to change the behavior of line-in + jack. Either "Line-In", "Rear Output" or "Bass Output" can + be selected. The last item is available only with model 039 + or newer. + When "Rear Output" is chosen, the surround channels 3 and 4 + are output to line-in jack. +"Mic-In Mode" - an enum control to change the behavior of mic-in + jack. Either "Mic-In" or "Center/LFE Output" can be + selected. + When "Center/LFE Output" is chosen, the center and bass + channels (channels 5 and 6) are output to mic-in jack. + +Digital I/O +----------- + +The CM8x38 provides the excellent SPDIF capability with very cheap +price (yes, that's the reason I bought the card :) + +The SPDIF playback and capture are done via the third PCM device +(hw:0,2). Usually this is assigned to the PCM device "spdif". +The available rates are 44100 and 48000 Hz. +For playback with aplay, you can run like below: + + % aplay -Dhw:0,2 foo.wav + +or + + % aplay -Dspdif foo.wav + +24bit format is also supported experimentally. + +The playback and capture over SPDIF use normal DAC and ADC, +respectively, so you cannot playback both analog and digital streams +simultaneously. + +To enable SPDIF output, you need to turn on "IEC958 Output Switch" +control via mixer or alsactl ("IEC958" is the official name of +so-called S/PDIF). Then you'll see the red light on from the card so +you know that's working obviously :) +The SPDIF input is always enabled, so you can hear SPDIF input data +from line-out with "IEC958 In Monitor" switch at any time (see +below). + +You can play via SPDIF even with the first device (hw:0,0), +but SPDIF is enabled only when the proper format (S16LE), sample rate +(441100 or 48000) and channels (2) are used. Otherwise it's turned +off. (Also don't forget to turn on "IEC958 Output Switch", too.) + + +Additionally there are relevant control switches: + +"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and + output through SPDIF. This switch appears only on old chip + models (CM8738 033 and 037). + Note: without this control you can output PCM to SPDIF. + This is "mixing" of streams, so e.g. it's not for AC3 output + (see the next section). + +"IEC958 In Select" - Select SPDIF input, the internal CD-in (false) + and the external input (true). + +"IEC958 Loop" - SPDIF input data is loop back into SPDIF + output (aka bypass) + +"IEC958 Copyright" - Set the copyright bit. + +"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface. + On some cards this doesn't work and you need to change the + configuration with hardware dip-switch. + +"IEC958 In Monitor" - SPDIF input is routed to DAC. + +"IEC958 In Phase Inverse" - Set SPDIF input format as inverse. + [FIXME: this doesn't work on all chips..] + +"IEC958 In Valid" - Set input validity flag detection. + +Note: When "PCM Playback Switch" is on, you'll hear the digital output +stream through analog line-out. + + +The AC3 (RAW DIGITAL) OUTPUT +---------------------------- + +The driver supports raw digital (typically AC3) i/o over SPDIF. This +can be toggled via IEC958 playback control, but usually you need to +access it via alsa-lib. See alsa-lib documents for more details. + +On the raw digital mode, the "PCM Playback Switch" is automatically +turned off so that non-audio data is heard from the analog line-out. +Similarly the following switches are off: "IEC958 Mix Analog" and +"IEC958 Loop". The switches are resumed after closing the SPDIF PCM +device automatically to the previous state. + +On the model 033, AC3 is implemented by the software conversion in +the alsa-lib. If you need to bypass the software conversion of IEC958 +subframes, pass the "soft_ac3=0" module option. This doesn't matter +on the newer models. + + +ANALOG MIXER INTERFACE +---------------------- + +The mixer interface on CM8x38 is similar to SB16. +There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback +volumes. Synth, CD, Line and Mic have playback and capture switches, +too, as well as SB16. + +In addition to the standard SB mixer, CM8x38 provides more functions. +- PCM playback switch +- PCM capture switch (to capture the data sent to DAC) +- Mic Boost switch +- Mic capture volume +- Aux playback volume/switch and capture switch +- 3D control switch + + +MIDI CONTROLLER +--------------- + +With CMI8338 chips, the MPU401-UART interface is disabled as default. +You need to set the module option "mpu_port" to a valid I/O port address +to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and +0x330. Choose a value that doesn't conflict with other cards. + +With CMI8738 and newer chips, the MIDI interface is enabled by default +and the driver automatically chooses a port address. + +There is _no_ hardware wavetable function on this chip (except for +OPL3 synth below). +What's said as MIDI synth on Windows is a software synthesizer +emulation. On Linux use TiMidity or other softsynth program for +playing MIDI music. + + +FM OPL/3 Synth +-------------- + +The FM OPL/3 is also enabled as default only for the first card. +Set "fm_port" module option for more cards. + +The output quality of FM OPL/3 is, however, very weird. +I don't know why.. + +CMI8768 and newer chips do not have the FM synth. + + +Joystick and Modem +------------------ + +The legacy joystick is supported. To enable the joystick support, pass +joystick_port=1 module option. The value 1 means the auto-detection. +If the auto-detection fails, try to pass the exact I/O address. + +The modem is enabled dynamically via a card control switch "Modem". + + +Debugging Information +--------------------- + +The registers are shown in /proc/asound/cardX/cmipci. If you have any +problem (especially unexpected behavior of mixer), please attach the +output of this proc file together with the bug report. diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt new file mode 100644 index 0000000..5b18298 --- /dev/null +++ b/Documentation/sound/alsa/ControlNames.txt @@ -0,0 +1,84 @@ +This document describes standard names of mixer controls. + +Syntax: SOURCE [DIRECTION] FUNCTION + +DIRECTION: + <nothing> (both directions) + Playback + Capture + Bypass Playback + Bypass Capture + +FUNCTION: + Switch (on/off switch) + Volume + Route (route control, hardware specific) + +SOURCE: + Master + Master Mono + Hardware Master + Headphone + PC Speaker + Phone + Phone Input + Phone Output + Synth + FM + Mic + Line + CD + Video + Zoom Video + Aux + PCM + PCM Front + PCM Rear + PCM Pan + Loopback + Analog Loopback (D/A -> A/D loopback) + Digital Loopback (playback -> capture loopback - without analog path) + Mono + Mono Output + Multi + ADC + Wave + Music + I2S + IEC958 + +Exceptions: + [Digital] Capture Source + [Digital] Capture Switch (aka input gain switch) + [Digital] Capture Volume (aka input gain volume) + [Digital] Playback Switch (aka output gain switch) + [Digital] Playback Volume (aka output gain volume) + Tone Control - Switch + Tone Control - Bass + Tone Control - Treble + 3D Control - Switch + 3D Control - Center + 3D Control - Depth + 3D Control - Wide + 3D Control - Space + 3D Control - Level + Mic Boost [(?dB)] + +PCM interface: + + Sample Clock Source { "Word", "Internal", "AutoSync" } + Clock Sync Status { "Lock", "Sync", "No Lock" } + External Rate /* external capture rate */ + Capture Rate /* capture rate taken from external source */ + +IEC958 (S/PDIF) interface: + + IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */ + IEC958 [...] [Playback|Capture] Volume /* digital volume control */ + IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */ + IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */ + IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */ + IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */ + IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */ + IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */ + IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */ diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl new file mode 100644 index 0000000..9d644f7 --- /dev/null +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -0,0 +1,100 @@ +<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook V4.1//EN"> + +<book> +<?dbhtml filename="index.html"> + +<!-- ****************************************************** --> +<!-- Header --> +<!-- ****************************************************** --> + <bookinfo> + <title>The ALSA Driver API</title> + + <legalnotice> + <para> + This document is free; 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. + </para> + + <para> + This document is distributed in the hope that it will be useful, + but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the + implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE</emphasis>. See the GNU General Public License + for more details. + </para> + + <para> + 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 + </para> + </legalnotice> + + </bookinfo> + + <chapter><title>Management of Cards and Devices</title> + <sect1><title>Card Management</title> +!Esound/core/init.c + </sect1> + <sect1><title>Device Components</title> +!Esound/core/device.c + </sect1> + <sect1><title>Module requests and Device File Entries</title> +!Esound/core/sound.c + </sect1> + <sect1><title>Memory Management Helpers</title> +!Esound/core/memory.c +!Esound/core/memalloc.c + </sect1> + </chapter> + <chapter><title>PCM API</title> + <sect1><title>PCM Core</title> +!Esound/core/pcm.c +!Esound/core/pcm_lib.c +!Esound/core/pcm_native.c + </sect1> + <sect1><title>PCM Format Helpers</title> +!Esound/core/pcm_misc.c + </sect1> + <sect1><title>PCM Memory Management</title> +!Esound/core/pcm_memory.c + </sect1> + </chapter> + <chapter><title>Control/Mixer API</title> + <sect1><title>General Control Interface</title> +!Esound/core/control.c + </sect1> + <sect1><title>AC97 Codec API</title> +!Esound/pci/ac97/ac97_codec.c +!Esound/pci/ac97/ac97_pcm.c + </sect1> + </chapter> + <chapter><title>MIDI API</title> + <sect1><title>Raw MIDI API</title> +!Esound/core/rawmidi.c + </sect1> + <sect1><title>MPU401-UART API</title> +!Esound/drivers/mpu401/mpu401_uart.c + </sect1> + </chapter> + <chapter><title>Proc Info API</title> + <sect1><title>Proc Info Interface</title> +!Esound/core/info.c + </sect1> + </chapter> + <chapter><title>Miscellaneous Functions</title> + <sect1><title>Hardware-Dependent Devices API</title> +!Esound/core/hwdep.c + </sect1> + <sect1><title>ISA DMA Helpers</title> +!Esound/core/isadma.c + </sect1> + <sect1><title>Other Helper Macros</title> +!Iinclude/sound/core.h + </sect1> + </chapter> + +</book> diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl new file mode 100644 index 0000000..87a7c07 --- /dev/null +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -0,0 +1,6210 @@ +<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook V4.1//EN"> + +<book> +<?dbhtml filename="index.html"> + +<!-- ****************************************************** --> +<!-- Header --> +<!-- ****************************************************** --> + <bookinfo> + <title>Writing an ALSA Driver</title> + <author> + <firstname>Takashi</firstname> + <surname>Iwai</surname> + <affiliation> + <address> + <email>tiwai@suse.de</email> + </address> + </affiliation> + </author> + + <date>Oct 15, 2007</date> + <edition>0.3.7</edition> + + <abstract> + <para> + This document describes how to write an ALSA (Advanced Linux + Sound Architecture) driver. + </para> + </abstract> + + <legalnotice> + <para> + Copyright (c) 2002-2005 Takashi Iwai <email>tiwai@suse.de</email> + </para> + + <para> + This document is free; 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. + </para> + + <para> + This document is distributed in the hope that it will be useful, + but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the + implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE</emphasis>. See the GNU General Public License + for more details. + </para> + + <para> + 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 + </para> + </legalnotice> + + </bookinfo> + +<!-- ****************************************************** --> +<!-- Preface --> +<!-- ****************************************************** --> + <preface id="preface"> + <title>Preface</title> + <para> + This document describes how to write an + <ulink url="http://www.alsa-project.org/"><citetitle> + ALSA (Advanced Linux Sound Architecture)</citetitle></ulink> + driver. The document focuses mainly on PCI soundcards. + In the case of other device types, the API might + be different, too. However, at least the ALSA kernel API is + consistent, and therefore it would be still a bit help for + writing them. + </para> + + <para> + This document targets people who already have enough + C language skills and have basic linux kernel programming + knowledge. This document doesn't explain the general + topic of linux kernel coding and doesn't cover low-level + driver implementation details. It only describes + the standard way to write a PCI sound driver on ALSA. + </para> + + <para> + If you are already familiar with the older ALSA ver.0.5.x API, you + can check the drivers such as <filename>sound/pci/es1938.c</filename> or + <filename>sound/pci/maestro3.c</filename> which have also almost the same + code-base in the ALSA 0.5.x tree, so you can compare the differences. + </para> + + <para> + This document is still a draft version. Any feedback and + corrections, please!! + </para> + </preface> + + +<!-- ****************************************************** --> +<!-- File Tree Structure --> +<!-- ****************************************************** --> + <chapter id="file-tree"> + <title>File Tree Structure</title> + + <section id="file-tree-general"> + <title>General</title> + <para> + The ALSA drivers are provided in two ways. + </para> + + <para> + One is the trees provided as a tarball or via cvs from the + ALSA's ftp site, and another is the 2.6 (or later) Linux kernel + tree. To synchronize both, the ALSA driver tree is split into + two different trees: alsa-kernel and alsa-driver. The former + contains purely the source code for the Linux 2.6 (or later) + tree. This tree is designed only for compilation on 2.6 or + later environment. The latter, alsa-driver, contains many subtle + files for compiling ALSA drivers outside of the Linux kernel tree, + wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, + and additional drivers which are still in development or in + tests. The drivers in alsa-driver tree will be moved to + alsa-kernel (and eventually to the 2.6 kernel tree) when they are + finished and confirmed to work fine. + </para> + + <para> + The file tree structure of ALSA driver is depicted below. Both + alsa-kernel and alsa-driver have almost the same file + structure, except for <quote>core</quote> directory. It's + named as <quote>acore</quote> in alsa-driver tree. + + <example> + <title>ALSA File Tree Structure</title> + <literallayout> + sound + /core + /oss + /seq + /oss + /instr + /ioctl32 + /include + /drivers + /mpu401 + /opl3 + /i2c + /l3 + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /oss + </literallayout> + </example> + </para> + </section> + + <section id="file-tree-core-directory"> + <title>core directory</title> + <para> + This directory contains the middle layer which is the heart + of ALSA drivers. In this directory, the native ALSA modules are + stored. The sub-directories contain different modules and are + dependent upon the kernel config. + </para> + + <section id="file-tree-core-directory-oss"> + <title>core/oss</title> + + <para> + The codes for PCM and mixer OSS emulation modules are stored + in this directory. The rawmidi OSS emulation is included in + the ALSA rawmidi code since it's quite small. The sequencer + code is stored in <filename>core/seq/oss</filename> directory (see + <link linkend="file-tree-core-directory-seq-oss"><citetitle> + below</citetitle></link>). + </para> + </section> + + <section id="file-tree-core-directory-ioctl32"> + <title>core/ioctl32</title> + + <para> + This directory contains the 32bit-ioctl wrappers for 64bit + architectures such like x86-64, ppc64 and sparc64. For 32bit + and alpha architectures, these are not compiled. + </para> + </section> + + <section id="file-tree-core-directory-seq"> + <title>core/seq</title> + <para> + This directory and its sub-directories are for the ALSA + sequencer. This directory contains the sequencer core and + primary sequencer modules such like snd-seq-midi, + snd-seq-virmidi, etc. They are compiled only when + <constant>CONFIG_SND_SEQUENCER</constant> is set in the kernel + config. + </para> + </section> + + <section id="file-tree-core-directory-seq-oss"> + <title>core/seq/oss</title> + <para> + This contains the OSS sequencer emulation codes. + </para> + </section> + + <section id="file-tree-core-directory-deq-instr"> + <title>core/seq/instr</title> + <para> + This directory contains the modules for the sequencer + instrument layer. + </para> + </section> + </section> + + <section id="file-tree-include-directory"> + <title>include directory</title> + <para> + This is the place for the public header files of ALSA drivers, + which are to be exported to user-space, or included by + several files at different directories. Basically, the private + header files should not be placed in this directory, but you may + still find files there, due to historical reasons :) + </para> + </section> + + <section id="file-tree-drivers-directory"> + <title>drivers directory</title> + <para> + This directory contains code shared among different drivers + on different architectures. They are hence supposed not to be + architecture-specific. + For example, the dummy pcm driver and the serial MIDI + driver are found in this directory. In the sub-directories, + there is code for components which are independent from + bus and cpu architectures. + </para> + + <section id="file-tree-drivers-directory-mpu401"> + <title>drivers/mpu401</title> + <para> + The MPU401 and MPU401-UART modules are stored here. + </para> + </section> + + <section id="file-tree-drivers-directory-opl3"> + <title>drivers/opl3 and opl4</title> + <para> + The OPL3 and OPL4 FM-synth stuff is found here. + </para> + </section> + </section> + + <section id="file-tree-i2c-directory"> + <title>i2c directory</title> + <para> + This contains the ALSA i2c components. + </para> + + <para> + Although there is a standard i2c layer on Linux, ALSA has its + own i2c code for some cards, because the soundcard needs only a + simple operation and the standard i2c API is too complicated for + such a purpose. + </para> + + <section id="file-tree-i2c-directory-l3"> + <title>i2c/l3</title> + <para> + This is a sub-directory for ARM L3 i2c. + </para> + </section> + </section> + + <section id="file-tree-synth-directory"> + <title>synth directory</title> + <para> + This contains the synth middle-level modules. + </para> + + <para> + So far, there is only Emu8000/Emu10k1 synth driver under + the <filename>synth/emux</filename> sub-directory. + </para> + </section> + + <section id="file-tree-pci-directory"> + <title>pci directory</title> + <para> + This directory and its sub-directories hold the top-level card modules + for PCI soundcards and the code specific to the PCI BUS. + </para> + + <para> + The drivers compiled from a single file are stored directly + in the pci directory, while the drivers with several source files are + stored on their own sub-directory (e.g. emu10k1, ice1712). + </para> + </section> + + <section id="file-tree-isa-directory"> + <title>isa directory</title> + <para> + This directory and its sub-directories hold the top-level card modules + for ISA soundcards. + </para> + </section> + + <section id="file-tree-arm-ppc-sparc-directories"> + <title>arm, ppc, and sparc directories</title> + <para> + They are used for top-level card modules which are + specific to one of these architectures. + </para> + </section> + + <section id="file-tree-usb-directory"> + <title>usb directory</title> + <para> + This directory contains the USB-audio driver. In the latest version, the + USB MIDI driver is integrated in the usb-audio driver. + </para> + </section> + + <section id="file-tree-pcmcia-directory"> + <title>pcmcia directory</title> + <para> + The PCMCIA, especially PCCard drivers will go here. CardBus + drivers will be in the pci directory, because their API is identical + to that of standard PCI cards. + </para> + </section> + + <section id="file-tree-oss-directory"> + <title>oss directory</title> + <para> + The OSS/Lite source files are stored here in Linux 2.6 (or + later) tree. In the ALSA driver tarball, this directory is empty, + of course :) + </para> + </section> + </chapter> + + +<!-- ****************************************************** --> +<!-- Basic Flow for PCI Drivers --> +<!-- ****************************************************** --> + <chapter id="basic-flow"> + <title>Basic Flow for PCI Drivers</title> + + <section id="basic-flow-outline"> + <title>Outline</title> + <para> + The minimum flow for PCI soundcards is as follows: + + <itemizedlist> + <listitem><para>define the PCI ID table (see the section + <link linkend="pci-resource-entries"><citetitle>PCI Entries + </citetitle></link>).</para></listitem> + <listitem><para>create <function>probe()</function> callback.</para></listitem> + <listitem><para>create <function>remove()</function> callback.</para></listitem> + <listitem><para>create a <structname>pci_driver</structname> structure + containing the three pointers above.</para></listitem> + <listitem><para>create an <function>init()</function> function just calling + the <function>pci_register_driver()</function> to register the pci_driver table + defined above.</para></listitem> + <listitem><para>create an <function>exit()</function> function to call + the <function>pci_unregister_driver()</function> function.</para></listitem> + </itemizedlist> + </para> + </section> + + <section id="basic-flow-example"> + <title>Full Code Example</title> + <para> + The code example is shown below. Some parts are kept + unimplemented at this moment but will be filled in the + next sections. The numbers in the comment lines of the + <function>snd_mychip_probe()</function> function + refer to details explained in the following section. + + <example> + <title>Basic Flow for PCI Drivers - Example</title> + <programlisting> +<![CDATA[ + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> + + /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + + /* definition of the chip-specific record */ + struct mychip { + struct snd_card *card; + /* the rest of the implementation will be in section + * "PCI Resource Management" + */ + }; + + /* chip-specific destructor + * (see "PCI Resource Management") + */ + static int snd_mychip_free(struct mychip *chip) + { + .... /* will be implemented later... */ + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int __devinit snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* check PCI availability here + * (see "PCI Resource Management") + */ + .... + + /* allocate a chip-specific data with zero filled */ + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + /* rest of initialization here; will be implemented + * later, see "PCI Resource Management" + */ + .... + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + return 0; + } + + /* constructor -- see "Constructor" sub-section */ + static int __devinit snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + static int dev; + struct snd_card *card; + struct mychip *chip; + int err; + + /* (1) */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + /* (3) */ + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see the "Destructor" sub-section */ + static void __devexit snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } +]]> + </programlisting> + </example> + </para> + </section> + + <section id="basic-flow-constructor"> + <title>Constructor</title> + <para> + The real constructor of PCI drivers is the <function>probe</function> callback. + The <function>probe</function> callback and other component-constructors which are called + from the <function>probe</function> callback should be defined with + the <parameter>__devinit</parameter> prefix. You + cannot use the <parameter>__init</parameter> prefix for them, + because any PCI device could be a hotplug device. + </para> + + <para> + In the <function>probe</function> callback, the following scheme is often used. + </para> + + <section id="basic-flow-constructor-device-index"> + <title>1) Check and increment the device index.</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int dev; + .... + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } +]]> + </programlisting> + </informalexample> + + where enable[dev] is the module option. + </para> + + <para> + Each time the <function>probe</function> callback is called, check the + availability of the device. If not available, simply increment + the device index and returns. dev will be incremented also + later (<link + linkend="basic-flow-constructor-set-pci"><citetitle>step + 7</citetitle></link>). + </para> + </section> + + <section id="basic-flow-constructor-create-card"> + <title>2) Create a card instance</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + struct snd_card *card; + .... + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The details will be explained in the section + <link linkend="card-management-card-instance"><citetitle> + Management of Cards and Components</citetitle></link>. + </para> + </section> + + <section id="basic-flow-constructor-create-main"> + <title>3) Create a main component</title> + <para> + In this part, the PCI resources are allocated. + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip *chip; + .... + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } +]]> + </programlisting> + </informalexample> + + The details will be explained in the section <link + linkend="pci-resource"><citetitle>PCI Resource + Management</citetitle></link>. + </para> + </section> + + <section id="basic-flow-constructor-main-component"> + <title>4) Set the driver ID and name strings.</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); +]]> + </programlisting> + </informalexample> + + The driver field holds the minimal ID string of the + chip. This is used by alsa-lib's configurator, so keep it + simple but unique. + Even the same driver can have different driver IDs to + distinguish the functionality of each chip type. + </para> + + <para> + The shortname field is a string shown as more verbose + name. The longname field contains the information + shown in <filename>/proc/asound/cards</filename>. + </para> + </section> + + <section id="basic-flow-constructor-create-other"> + <title>5) Create other components, such as mixer, MIDI, etc.</title> + <para> + Here you define the basic components such as + <link linkend="pcm-interface"><citetitle>PCM</citetitle></link>, + mixer (e.g. <link linkend="api-ac97"><citetitle>AC97</citetitle></link>), + MIDI (e.g. <link linkend="midi-interface"><citetitle>MPU-401</citetitle></link>), + and other interfaces. + Also, if you want a <link linkend="proc-interface"><citetitle>proc + file</citetitle></link>, define it here, too. + </para> + </section> + + <section id="basic-flow-constructor-register-card"> + <title>6) Register the card instance.</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + Will be explained in the section <link + linkend="card-management-registration"><citetitle>Management + of Cards and Components</citetitle></link>, too. + </para> + </section> + + <section id="basic-flow-constructor-set-pci"> + <title>7) Set the PCI driver data and return zero.</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + pci_set_drvdata(pci, card); + dev++; + return 0; +]]> + </programlisting> + </informalexample> + + In the above, the card record is stored. This pointer is + used in the remove callback and power-management + callbacks, too. + </para> + </section> + </section> + + <section id="basic-flow-destructor"> + <title>Destructor</title> + <para> + The destructor, remove callback, simply releases the card + instance. Then the ALSA middle layer will release all the + attached components automatically. + </para> + + <para> + It would be typically like the following: + + <informalexample> + <programlisting> +<![CDATA[ + static void __devexit snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } +]]> + </programlisting> + </informalexample> + + The above code assumes that the card pointer is set to the PCI + driver data. + </para> + </section> + + <section id="basic-flow-header-files"> + <title>Header Files</title> + <para> + For the above example, at least the following include files + are necessary. + + <informalexample> + <programlisting> +<![CDATA[ + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> +]]> + </programlisting> + </informalexample> + + where the last one is necessary only when module options are + defined in the source file. If the code is split into several + files, the files without module options don't need them. + </para> + + <para> + In addition to these headers, you'll need + <filename><linux/interrupt.h></filename> for interrupt + handling, and <filename><asm/io.h></filename> for I/O + access. If you use the <function>mdelay()</function> or + <function>udelay()</function> functions, you'll need to include + <filename><linux/delay.h></filename> too. + </para> + + <para> + The ALSA interfaces like the PCM and control APIs are defined in other + <filename><sound/xxx.h></filename> header files. + They have to be included after + <filename><sound/core.h></filename>. + </para> + + </section> + </chapter> + + +<!-- ****************************************************** --> +<!-- Management of Cards and Components --> +<!-- ****************************************************** --> + <chapter id="card-management"> + <title>Management of Cards and Components</title> + + <section id="card-management-card-instance"> + <title>Card Instance</title> + <para> + For each soundcard, a <quote>card</quote> record must be allocated. + </para> + + <para> + A card record is the headquarters of the soundcard. It manages + the whole list of devices (components) on the soundcard, such as + PCM, mixers, MIDI, synthesizer, and so on. Also, the card + record holds the ID and the name strings of the card, manages + the root of proc files, and controls the power-management states + and hotplug disconnections. The component list on the card + record is used to manage the correct release of resources at + destruction. + </para> + + <para> + As mentioned above, to create a card instance, call + <function>snd_card_new()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_card *card; + card = snd_card_new(index, id, module, extra_size); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The function takes four arguments, the card-index number, the + id string, the module pointer (usually + <constant>THIS_MODULE</constant>), + and the size of extra-data space. The last argument is used to + allocate card->private_data for the + chip-specific data. Note that these data + are allocated by <function>snd_card_new()</function>. + </para> + </section> + + <section id="card-management-component"> + <title>Components</title> + <para> + After the card is created, you can attach the components + (devices) to the card instance. In an ALSA driver, a component is + represented as a struct <structname>snd_device</structname> object. + A component can be a PCM instance, a control interface, a raw + MIDI interface, etc. Each such instance has one component + entry. + </para> + + <para> + A component can be created via + <function>snd_device_new()</function> function. + + <informalexample> + <programlisting> +<![CDATA[ + snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); +]]> + </programlisting> + </informalexample> + </para> + + <para> + This takes the card pointer, the device-level + (<constant>SNDRV_DEV_XXX</constant>), the data pointer, and the + callback pointers (<parameter>&ops</parameter>). The + device-level defines the type of components and the order of + registration and de-registration. For most components, the + device-level is already defined. For a user-defined component, + you can use <constant>SNDRV_DEV_LOWLEVEL</constant>. + </para> + + <para> + This function itself doesn't allocate the data space. The data + must be allocated manually beforehand, and its pointer is passed + as the argument. This pointer is used as the + (<parameter>chip</parameter> identifier in the above example) + for the instance. + </para> + + <para> + Each pre-defined ALSA component such as ac97 and pcm calls + <function>snd_device_new()</function> inside its + constructor. The destructor for each component is defined in the + callback pointers. Hence, you don't need to take care of + calling a destructor for such a component. + </para> + + <para> + If you wish to create your own component, you need to + set the destructor function to the dev_free callback in + the <parameter>ops</parameter>, so that it can be released + automatically via <function>snd_card_free()</function>. + The next example will show an implementation of chip-specific + data. + </para> + </section> + + <section id="card-management-chip-specific"> + <title>Chip-Specific Data</title> + <para> + Chip-specific information, e.g. the I/O port address, its + resource pointer, or the irq number, is stored in the + chip-specific record. + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip { + .... + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + In general, there are two ways of allocating the chip record. + </para> + + <section id="card-management-chip-specific-snd-card-new"> + <title>1. Allocating via <function>snd_card_new()</function>.</title> + <para> + As mentioned above, you can pass the extra-data-length + to the 4th argument of <function>snd_card_new()</function>, i.e. + + <informalexample> + <programlisting> +<![CDATA[ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct mychip)); +]]> + </programlisting> + </informalexample> + + struct <structname>mychip</structname> is the type of the chip record. + </para> + + <para> + In return, the allocated record can be accessed as + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip *chip = card->private_data; +]]> + </programlisting> + </informalexample> + + With this method, you don't have to allocate twice. + The record is released together with the card instance. + </para> + </section> + + <section id="card-management-chip-specific-allocate-extra"> + <title>2. Allocating an extra device.</title> + + <para> + After allocating a card instance via + <function>snd_card_new()</function> (with + <constant>NULL</constant> on the 4th arg), call + <function>kzalloc()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_card *card; + struct mychip *chip; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, NULL); + ..... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The chip record should have the field to hold the card + pointer at least, + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip { + struct snd_card *card; + .... + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + Then, set the card pointer in the returned chip instance. + + <informalexample> + <programlisting> +<![CDATA[ + chip->card = card; +]]> + </programlisting> + </informalexample> + </para> + + <para> + Next, initialize the fields, and register this chip + record as a low-level device with a specified + <parameter>ops</parameter>, + + <informalexample> + <programlisting> +<![CDATA[ + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + .... + snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); +]]> + </programlisting> + </informalexample> + + <function>snd_mychip_dev_free()</function> is the + device-destructor function, which will call the real + destructor. + </para> + + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } +]]> + </programlisting> + </informalexample> + + where <function>snd_mychip_free()</function> is the real destructor. + </para> + </section> + </section> + + <section id="card-management-registration"> + <title>Registration and Release</title> + <para> + After all components are assigned, register the card instance + by calling <function>snd_card_register()</function>. Access + to the device files is enabled at this point. That is, before + <function>snd_card_register()</function> is called, the + components are safely inaccessible from external side. If this + call fails, exit the probe function after releasing the card via + <function>snd_card_free()</function>. + </para> + + <para> + For releasing the card instance, you can call simply + <function>snd_card_free()</function>. As mentioned earlier, all + components are released automatically by this call. + </para> + + <para> + As further notes, the destructors (both + <function>snd_mychip_dev_free</function> and + <function>snd_mychip_free</function>) cannot be defined with + the <parameter>__devexit</parameter> prefix, because they may be + called from the constructor, too, at the false path. + </para> + + <para> + For a device which allows hotplugging, you can use + <function>snd_card_free_when_closed</function>. This one will + postpone the destruction until all devices are closed. + </para> + + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- PCI Resource Management --> +<!-- ****************************************************** --> + <chapter id="pci-resource"> + <title>PCI Resource Management</title> + + <section id="pci-resource-example"> + <title>Full Code Example</title> + <para> + In this section, we'll complete the chip-specific constructor, + destructor and PCI entries. Example code is shown first, + below. + + <example> + <title>PCI Resource Management Example</title> + <programlisting> +<![CDATA[ + struct mychip { + struct snd_card *card; + struct pci_dev *pci; + + unsigned long port; + int irq; + }; + + static int snd_mychip_free(struct mychip *chip) + { + /* disable hardware here if any */ + .... /* (not implemented in this document) */ + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, chip); + /* release the I/O ports & memory */ + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; + } + + /* chip-specific constructor */ + static int __devinit snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, "My Chip", chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + .... /* (not implemented in this document) */ + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + return 0; + } + + /* PCI IDs */ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + + /* pci_driver definition */ + static struct pci_driver driver = { + .name = "My Own Chip", + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = __devexit_p(snd_mychip_remove), + }; + + /* module initialization */ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + /* module clean up */ + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + + EXPORT_NO_SYMBOLS; /* for old kernels only */ +]]> + </programlisting> + </example> + </para> + </section> + + <section id="pci-resource-some-haftas"> + <title>Some Hafta's</title> + <para> + The allocation of PCI resources is done in the + <function>probe()</function> function, and usually an extra + <function>xxx_create()</function> function is written for this + purpose. + </para> + + <para> + In the case of PCI devices, you first have to call + the <function>pci_enable_device()</function> function before + allocating resources. Also, you need to set the proper PCI DMA + mask to limit the accessed I/O range. In some cases, you might + need to call <function>pci_set_master()</function> function, + too. + </para> + + <para> + Suppose the 28bit mask, and the code to be added would be like: + + <informalexample> + <programlisting> +<![CDATA[ + err = pci_enable_device(pci); + if (err < 0) + return err; + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + +]]> + </programlisting> + </informalexample> + </para> + </section> + + <section id="pci-resource-resource-allocation"> + <title>Resource Allocation</title> + <para> + The allocation of I/O ports and irqs is done via standard kernel + functions. Unlike ALSA ver.0.5.x., there are no helpers for + that. And these resources must be released in the destructor + function (see below). Also, on ALSA 0.9.x, you don't need to + allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. + </para> + + <para> + Now assume that the PCI device has an I/O port with 8 bytes + and an interrupt. Then struct <structname>mychip</structname> will have the + following fields: + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip { + struct snd_card *card; + + unsigned long port; + int irq; + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + For an I/O port (and also a memory region), you need to have + the resource pointer for the standard resource management. For + an irq, you have to keep only the irq number (integer). But you + need to initialize this number as -1 before actual allocation, + since irq 0 is valid. The port address and its resource pointer + can be initialized as null by + <function>kzalloc()</function> automatically, so you + don't have to take care of resetting them. + </para> + + <para> + The allocation of an I/O port is done like this: + + <informalexample> + <programlisting> +<![CDATA[ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); +]]> + </programlisting> + </informalexample> + </para> + + <para> + <!-- obsolete --> + It will reserve the I/O port region of 8 bytes of the given + PCI device. The returned value, chip->res_port, is allocated + via <function>kmalloc()</function> by + <function>request_region()</function>. The pointer must be + released via <function>kfree()</function>, but there is a + problem with this. This issue will be explained later. + </para> + + <para> + The allocation of an interrupt source is done like this: + + <informalexample> + <programlisting> +<![CDATA[ + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, "My Chip", chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; +]]> + </programlisting> + </informalexample> + + where <function>snd_mychip_interrupt()</function> is the + interrupt handler defined <link + linkend="pcm-interface-interrupt-handler"><citetitle>later</citetitle></link>. + Note that chip->irq should be defined + only when <function>request_irq()</function> succeeded. + </para> + + <para> + On the PCI bus, interrupts can be shared. Thus, + <constant>IRQF_SHARED</constant> is used as the interrupt flag of + <function>request_irq()</function>. + </para> + + <para> + The last argument of <function>request_irq()</function> is the + data pointer passed to the interrupt handler. Usually, the + chip-specific record is used for that, but you can use what you + like, too. + </para> + + <para> + I won't give details about the interrupt handler at this + point, but at least its appearance can be explained now. The + interrupt handler looks usually like the following: + + <informalexample> + <programlisting> +<![CDATA[ + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + .... + return IRQ_HANDLED; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + Now let's write the corresponding destructor for the resources + above. The role of destructor is simple: disable the hardware + (if already activated) and release the resources. So far, we + have no hardware part, so the disabling code is not written here. + </para> + + <para> + To release the resources, the <quote>check-and-release</quote> + method is a safer way. For the interrupt, do like this: + + <informalexample> + <programlisting> +<![CDATA[ + if (chip->irq >= 0) + free_irq(chip->irq, chip); +]]> + </programlisting> + </informalexample> + + Since the irq number can start from 0, you should initialize + chip->irq with a negative value (e.g. -1), so that you can + check the validity of the irq number as above. + </para> + + <para> + When you requested I/O ports or memory regions via + <function>pci_request_region()</function> or + <function>pci_request_regions()</function> like in this example, + release the resource(s) using the corresponding function, + <function>pci_release_region()</function> or + <function>pci_release_regions()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + pci_release_regions(chip->pci); +]]> + </programlisting> + </informalexample> + </para> + + <para> + When you requested manually via <function>request_region()</function> + or <function>request_mem_region</function>, you can release it via + <function>release_resource()</function>. Suppose that you keep + the resource pointer returned from <function>request_region()</function> + in chip->res_port, the release procedure looks like: + + <informalexample> + <programlisting> +<![CDATA[ + release_and_free_resource(chip->res_port); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Don't forget to call <function>pci_disable_device()</function> + before the end. + </para> + + <para> + And finally, release the chip-specific record. + + <informalexample> + <programlisting> +<![CDATA[ + kfree(chip); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Again, remember that you cannot + use the <parameter>__devexit</parameter> prefix for this destructor. + </para> + + <para> + We didn't implement the hardware disabling part in the above. + If you need to do this, please note that the destructor may be + called even before the initialization of the chip is completed. + It would be better to have a flag to skip hardware disabling + if the hardware was not initialized yet. + </para> + + <para> + When the chip-data is assigned to the card using + <function>snd_device_new()</function> with + <constant>SNDRV_DEV_LOWLELVEL</constant> , its destructor is + called at the last. That is, it is assured that all other + components like PCMs and controls have already been released. + You don't have to stop PCMs, etc. explicitly, but just + call low-level hardware stopping. + </para> + + <para> + The management of a memory-mapped region is almost as same as + the management of an I/O port. You'll need three fields like + the following: + + <informalexample> + <programlisting> +<![CDATA[ + struct mychip { + .... + unsigned long iobase_phys; + void __iomem *iobase_virt; + }; +]]> + </programlisting> + </informalexample> + + and the allocation would be like below: + + <informalexample> + <programlisting> +<![CDATA[ + if ((err = pci_request_regions(pci, "My Chip")) < 0) { + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); +]]> + </programlisting> + </informalexample> + + and the corresponding destructor would be: + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_mychip_free(struct mychip *chip) + { + .... + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } +]]> + </programlisting> + </informalexample> + </para> + + </section> + + <section id="pci-resource-device-struct"> + <title>Registration of Device Struct</title> + <para> + At some point, typically after calling <function>snd_device_new()</function>, + you need to register the struct <structname>device</structname> of the chip + you're handling for udev and co. ALSA provides a macro for compatibility with + older kernels. Simply call like the following: + <informalexample> + <programlisting> +<![CDATA[ + snd_card_set_dev(card, &pci->dev); +]]> + </programlisting> + </informalexample> + so that it stores the PCI's device pointer to the card. This will be + referred by ALSA core functions later when the devices are registered. + </para> + <para> + In the case of non-PCI, pass the proper device struct pointer of the BUS + instead. (In the case of legacy ISA without PnP, you don't have to do + anything.) + </para> + </section> + + <section id="pci-resource-entries"> + <title>PCI Entries</title> + <para> + So far, so good. Let's finish the missing PCI + stuff. At first, we need a + <structname>pci_device_id</structname> table for this + chipset. It's a table of PCI vendor/device ID number, and some + masks. + </para> + + <para> + For example, + + <informalexample> + <programlisting> +<![CDATA[ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The first and second fields of + the <structname>pci_device_id</structname> structure are the vendor and + device IDs. If you have no reason to filter the matching + devices, you can leave the remaining fields as above. The last + field of the <structname>pci_device_id</structname> struct contains + private data for this entry. You can specify any value here, for + example, to define specific operations for supported device IDs. + Such an example is found in the intel8x0 driver. + </para> + + <para> + The last entry of this list is the terminator. You must + specify this all-zero entry. + </para> + + <para> + Then, prepare the <structname>pci_driver</structname> record: + + <informalexample> + <programlisting> +<![CDATA[ + static struct pci_driver driver = { + .name = "My Own Chip", + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = __devexit_p(snd_mychip_remove), + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The <structfield>probe</structfield> and + <structfield>remove</structfield> functions have already + been defined in the previous sections. + The <structfield>remove</structfield> function should + be defined with the + <function>__devexit_p()</function> macro, so that it's not + defined for built-in (and non-hot-pluggable) case. The + <structfield>name</structfield> + field is the name string of this device. Note that you must not + use a slash <quote>/</quote> in this string. + </para> + + <para> + And at last, the module entries: + + <informalexample> + <programlisting> +<![CDATA[ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) +]]> + </programlisting> + </informalexample> + </para> + + <para> + Note that these module entries are tagged with + <parameter>__init</parameter> and + <parameter>__exit</parameter> prefixes, not + <parameter>__devinit</parameter> nor + <parameter>__devexit</parameter>. + </para> + + <para> + Oh, one thing was forgotten. If you have no exported symbols, + you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). + + <informalexample> + <programlisting> +<![CDATA[ + EXPORT_NO_SYMBOLS; +]]> + </programlisting> + </informalexample> + + That's all! + </para> + </section> + </chapter> + + +<!-- ****************************************************** --> +<!-- PCM Interface --> +<!-- ****************************************************** --> + <chapter id="pcm-interface"> + <title>PCM Interface</title> + + <section id="pcm-interface-general"> + <title>General</title> + <para> + The PCM middle layer of ALSA is quite powerful and it is only + necessary for each driver to implement the low-level functions + to access its hardware. + </para> + + <para> + For accessing to the PCM layer, you need to include + <filename><sound/pcm.h></filename> first. In addition, + <filename><sound/pcm_params.h></filename> might be needed + if you access to some functions related with hw_param. + </para> + + <para> + Each card device can have up to four pcm instances. A pcm + instance corresponds to a pcm device file. The limitation of + number of instances comes only from the available bit size of + the Linux's device numbers. Once when 64bit device number is + used, we'll have more pcm instances available. + </para> + + <para> + A pcm instance consists of pcm playback and capture streams, + and each pcm stream consists of one or more pcm substreams. Some + soundcards support multiple playback functions. For example, + emu10k1 has a PCM playback of 32 stereo substreams. In this case, at + each open, a free substream is (usually) automatically chosen + and opened. Meanwhile, when only one substream exists and it was + already opened, the successful open will either block + or error with <constant>EAGAIN</constant> according to the + file open mode. But you don't have to care about such details in your + driver. The PCM middle layer will take care of such work. + </para> + </section> + + <section id="pcm-interface-example"> + <title>Full Code Example</title> + <para> + The example code below does not include any hardware access + routines but shows only the skeleton, how to build up the PCM + interfaces. + + <example> + <title>PCM Example Code</title> + <programlisting> +<![CDATA[ + #include <sound/pcm.h> + .... + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* open callback */ + static int snd_mychip_playback_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_playback_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_capture_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_capture_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* hw_params callback */ + static int snd_mychip_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)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int __devinit snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + return 0; + } +]]> + </programlisting> + </example> + </para> + </section> + + <section id="pcm-interface-constructor"> + <title>Constructor</title> + <para> + A pcm instance is allocated by the <function>snd_pcm_new()</function> + function. It would be better to create a constructor for pcm, + namely, + + <informalexample> + <programlisting> +<![CDATA[ + static int __devinit snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + The <function>snd_pcm_new()</function> function takes four + arguments. The first argument is the card pointer to which this + pcm is assigned, and the second is the ID string. + </para> + + <para> + The third argument (<parameter>index</parameter>, 0 in the + above) is the index of this new pcm. It begins from zero. If + you create more than one pcm instances, specify the + different numbers in this argument. For example, + <parameter>index</parameter> = 1 for the second PCM device. + </para> + + <para> + The fourth and fifth arguments are the number of substreams + for playback and capture, respectively. Here 1 is used for + both arguments. When no playback or capture substreams are available, + pass 0 to the corresponding argument. + </para> + + <para> + If a chip supports multiple playbacks or captures, you can + specify more numbers, but they must be handled properly in + open/close, etc. callbacks. When you need to know which + substream you are referring to, then it can be obtained from + struct <structname>snd_pcm_substream</structname> data passed to each callback + as follows: + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_pcm_substream *substream; + int index = substream->number; +]]> + </programlisting> + </informalexample> + </para> + + <para> + After the pcm is created, you need to set operators for each + pcm stream. + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The operators are defined typically like this: + + <informalexample> + <programlisting> +<![CDATA[ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_pcm_open, + .close = snd_mychip_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; +]]> + </programlisting> + </informalexample> + + All the callbacks are described in the + <link linkend="pcm-interface-operators"><citetitle> + Operators</citetitle></link> subsection. + </para> + + <para> + After setting the operators, you probably will want to + pre-allocate the buffer. For the pre-allocation, simply call + the following: + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); +]]> + </programlisting> + </informalexample> + + It will allocate a buffer up to 64kB as default. + Buffer management details will be described in the later section <link + linkend="buffer-and-memory"><citetitle>Buffer and Memory + Management</citetitle></link>. + </para> + + <para> + Additionally, you can set some extra information for this pcm + in pcm->info_flags. + The available values are defined as + <constant>SNDRV_PCM_INFO_XXX</constant> in + <filename><sound/asound.h></filename>, which is used for + the hardware definition (described later). When your soundchip + supports only half-duplex, specify like this: + + <informalexample> + <programlisting> +<![CDATA[ + pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; +]]> + </programlisting> + </informalexample> + </para> + </section> + + <section id="pcm-interface-destructor"> + <title>... And the Destructor?</title> + <para> + The destructor for a pcm instance is not always + necessary. Since the pcm device will be released by the middle + layer code automatically, you don't have to call the destructor + explicitly. + </para> + + <para> + The destructor would be necessary if you created + special records internally and needed to release them. In such a + case, set the destructor function to + pcm->private_free: + + <example> + <title>PCM Instance with a Destructor</title> + <programlisting> +<![CDATA[ + static void mychip_pcm_free(struct snd_pcm *pcm) + { + struct mychip *chip = snd_pcm_chip(pcm); + /* free your own data */ + kfree(chip->my_private_pcm_data); + /* do what you like else */ + .... + } + + static int __devinit snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } +]]> + </programlisting> + </example> + </para> + </section> + + <section id="pcm-interface-runtime"> + <title>Runtime Pointer - The Chest of PCM Information</title> + <para> + When the PCM substream is opened, a PCM runtime instance is + allocated and assigned to the substream. This pointer is + accessible via <constant>substream->runtime</constant>. + This runtime pointer holds most information you need + to control the PCM: the copy of hw_params and sw_params configurations, the buffer + pointers, mmap records, spinlocks, etc. + </para> + + <para> + The definition of runtime instance is found in + <filename><sound/pcm.h></filename>. Here are + the contents of this file: + <informalexample> + <programlisting> +<![CDATA[ +struct _snd_pcm_runtime { + /* -- Status -- */ + struct snd_pcm_substream *trigger_master; + snd_timestamp_t trigger_tstamp; /* trigger timestamp */ + int overrange; + snd_pcm_uframes_t avail_max; + snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + + /* -- HW params -- */ + snd_pcm_access_t access; /* access mode */ + snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ + snd_pcm_subformat_t subformat; /* subformat */ + unsigned int rate; /* rate in Hz */ + unsigned int channels; /* channels */ + snd_pcm_uframes_t period_size; /* period size */ + unsigned int periods; /* periods */ + snd_pcm_uframes_t buffer_size; /* buffer size */ + unsigned int tick_time; /* tick time */ + snd_pcm_uframes_t min_align; /* Min alignment for the format */ + size_t byte_align; + unsigned int frame_bits; + unsigned int sample_bits; + unsigned int info; + unsigned int rate_num; + unsigned int rate_den; + + /* -- SW params -- */ + struct timespec tstamp_mode; /* mmap timestamp is updated */ + unsigned int period_step; + unsigned int sleep_min; /* min ticks to sleep */ + snd_pcm_uframes_t start_threshold; + snd_pcm_uframes_t stop_threshold; + snd_pcm_uframes_t silence_threshold; /* Silence filling happens when + noise is nearest than this */ + snd_pcm_uframes_t silence_size; /* Silence filling size */ + snd_pcm_uframes_t boundary; /* pointers wrap point */ + + snd_pcm_uframes_t silenced_start; + snd_pcm_uframes_t silenced_size; + + snd_pcm_sync_id_t sync; /* hardware synchronization ID */ + + /* -- mmap -- */ + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + atomic_t mmap_count; + + /* -- locking / scheduling -- */ + spinlock_t lock; + wait_queue_head_t sleep; + struct timer_list tick_timer; + struct fasync_struct *fasync; + + /* -- private section -- */ + void *private_data; + void (*private_free)(struct snd_pcm_runtime *runtime); + + /* -- hardware description -- */ + struct snd_pcm_hardware hw; + struct snd_pcm_hw_constraints hw_constraints; + + /* -- interrupt callbacks -- */ + void (*transfer_ack_begin)(struct snd_pcm_substream *substream); + void (*transfer_ack_end)(struct snd_pcm_substream *substream); + + /* -- timer -- */ + unsigned int timer_resolution; /* timer resolution */ + + /* -- DMA -- */ + unsigned char *dma_area; /* DMA area */ + dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ + size_t dma_bytes; /* size of DMA area */ + + struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + +#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) + /* -- OSS things -- */ + struct snd_pcm_oss_runtime oss; +#endif +}; +]]> + </programlisting> + </informalexample> + </para> + + <para> + For the operators (callbacks) of each sound driver, most of + these records are supposed to be read-only. Only the PCM + middle-layer changes / updates them. The exceptions are + the hardware description (hw), interrupt callbacks + (transfer_ack_xxx), DMA buffer information, and the private + data. Besides, if you use the standard buffer allocation + method via <function>snd_pcm_lib_malloc_pages()</function>, + you don't need to set the DMA buffer information by yourself. + </para> + + <para> + In the sections below, important records are explained. + </para> + + <section id="pcm-interface-runtime-hw"> + <title>Hardware Description</title> + <para> + The hardware descriptor (struct <structname>snd_pcm_hardware</structname>) + contains the definitions of the fundamental hardware + configuration. Above all, you'll need to define this in + <link linkend="pcm-interface-operators-open-callback"><citetitle> + the open callback</citetitle></link>. + Note that the runtime instance holds the copy of the + descriptor, not the pointer to the existing descriptor. That + is, in the open callback, you can modify the copied descriptor + (<constant>runtime->hw</constant>) as you need. For example, if the maximum + number of channels is 1 only on some chip models, you can + still use the same hardware descriptor and change the + channels_max later: + <informalexample> + <programlisting> +<![CDATA[ + struct snd_pcm_runtime *runtime = substream->runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; +]]> + </programlisting> + </informalexample> + </para> + + <para> + Typically, you'll have a hardware descriptor as below: + <informalexample> + <programlisting> +<![CDATA[ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + <itemizedlist> + <listitem><para> + The <structfield>info</structfield> field contains the type and + capabilities of this pcm. The bit flags are defined in + <filename><sound/asound.h></filename> as + <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you + have to specify whether the mmap is supported and which + interleaved format is supported. + When the is supported, add the + <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the + hardware supports the interleaved or the non-interleaved + formats, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or + <constant>SNDRV_PCM_INFO_NONINTERLEAVED</constant> flag must + be set, respectively. If both are supported, you can set both, + too. + </para> + + <para> + In the above example, <constant>MMAP_VALID</constant> and + <constant>BLOCK_TRANSFER</constant> are specified for the OSS mmap + mode. Usually both are set. Of course, + <constant>MMAP_VALID</constant> is set only if the mmap is + really supported. + </para> + + <para> + The other possible flags are + <constant>SNDRV_PCM_INFO_PAUSE</constant> and + <constant>SNDRV_PCM_INFO_RESUME</constant>. The + <constant>PAUSE</constant> bit means that the pcm supports the + <quote>pause</quote> operation, while the + <constant>RESUME</constant> bit means that the pcm supports + the full <quote>suspend/resume</quote> operation. + If the <constant>PAUSE</constant> flag is set, + the <structfield>trigger</structfield> callback below + must handle the corresponding (pause push/release) commands. + The suspend/resume trigger commands can be defined even without + the <constant>RESUME</constant> flag. See <link + linkend="power-management"><citetitle> + Power Management</citetitle></link> section for details. + </para> + + <para> + When the PCM substreams can be synchronized (typically, + synchronized start/stop of a playback and a capture streams), + you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>, + too. In this case, you'll need to check the linked-list of + PCM substreams in the trigger callback. This will be + described in the later section. + </para> + </listitem> + + <listitem> + <para> + <structfield>formats</structfield> field contains the bit-flags + of supported formats (<constant>SNDRV_PCM_FMTBIT_XXX</constant>). + If the hardware supports more than one format, give all or'ed + bits. In the example above, the signed 16bit little-endian + format is specified. + </para> + </listitem> + + <listitem> + <para> + <structfield>rates</structfield> field contains the bit-flags of + supported rates (<constant>SNDRV_PCM_RATE_XXX</constant>). + When the chip supports continuous rates, pass + <constant>CONTINUOUS</constant> bit additionally. + The pre-defined rate bits are provided only for typical + rates. If your chip supports unconventional rates, you need to add + the <constant>KNOT</constant> bit and set up the hardware + constraint manually (explained later). + </para> + </listitem> + + <listitem> + <para> + <structfield>rate_min</structfield> and + <structfield>rate_max</structfield> define the minimum and + maximum sample rate. This should correspond somehow to + <structfield>rates</structfield> bits. + </para> + </listitem> + + <listitem> + <para> + <structfield>channel_min</structfield> and + <structfield>channel_max</structfield> + define, as you might already expected, the minimum and maximum + number of channels. + </para> + </listitem> + + <listitem> + <para> + <structfield>buffer_bytes_max</structfield> defines the + maximum buffer size in bytes. There is no + <structfield>buffer_bytes_min</structfield> field, since + it can be calculated from the minimum period size and the + minimum number of periods. + Meanwhile, <structfield>period_bytes_min</structfield> and + define the minimum and maximum size of the period in bytes. + <structfield>periods_max</structfield> and + <structfield>periods_min</structfield> define the maximum and + minimum number of periods in the buffer. + </para> + + <para> + The <quote>period</quote> is a term that corresponds to + a fragment in the OSS world. The period defines the size at + which a PCM interrupt is generated. This size strongly + depends on the hardware. + Generally, the smaller period size will give you more + interrupts, that is, more controls. + In the case of capture, this size defines the input latency. + On the other hand, the whole buffer size defines the + output latency for the playback direction. + </para> + </listitem> + + <listitem> + <para> + There is also a field <structfield>fifo_size</structfield>. + This specifies the size of the hardware FIFO, but currently it + is neither used in the driver nor in the alsa-lib. So, you + can ignore this field. + </para> + </listitem> + </itemizedlist> + </para> + </section> + + <section id="pcm-interface-runtime-config"> + <title>PCM Configurations</title> + <para> + Ok, let's go back again to the PCM runtime records. + The most frequently referred records in the runtime instance are + the PCM configurations. + The PCM configurations are stored in the runtime instance + after the application sends <type>hw_params</type> data via + alsa-lib. There are many fields copied from hw_params and + sw_params structs. For example, + <structfield>format</structfield> holds the format type + chosen by the application. This field contains the enum value + <constant>SNDRV_PCM_FORMAT_XXX</constant>. + </para> + + <para> + One thing to be noted is that the configured buffer and period + sizes are stored in <quote>frames</quote> in the runtime. + In the ALSA world, 1 frame = channels * samples-size. + For conversion between frames and bytes, you can use the + <function>frames_to_bytes()</function> and + <function>bytes_to_frames()</function> helper functions. + <informalexample> + <programlisting> +<![CDATA[ + period_bytes = frames_to_bytes(runtime, runtime->period_size); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Also, many software parameters (sw_params) are + stored in frames, too. Please check the type of the field. + <type>snd_pcm_uframes_t</type> is for the frames as unsigned + integer while <type>snd_pcm_sframes_t</type> is for the frames + as signed integer. + </para> + </section> + + <section id="pcm-interface-runtime-dma"> + <title>DMA Buffer Information</title> + <para> + The DMA buffer is defined by the following four fields, + <structfield>dma_area</structfield>, + <structfield>dma_addr</structfield>, + <structfield>dma_bytes</structfield> and + <structfield>dma_private</structfield>. + The <structfield>dma_area</structfield> holds the buffer + pointer (the logical address). You can call + <function>memcpy</function> from/to + this pointer. Meanwhile, <structfield>dma_addr</structfield> + holds the physical address of the buffer. This field is + specified only when the buffer is a linear buffer. + <structfield>dma_bytes</structfield> holds the size of buffer + in bytes. <structfield>dma_private</structfield> is used for + the ALSA DMA allocator. + </para> + + <para> + If you use a standard ALSA function, + <function>snd_pcm_lib_malloc_pages()</function>, for + allocating the buffer, these fields are set by the ALSA middle + layer, and you should <emphasis>not</emphasis> change them by + yourself. You can read them but not write them. + On the other hand, if you want to allocate the buffer by + yourself, you'll need to manage it in hw_params callback. + At least, <structfield>dma_bytes</structfield> is mandatory. + <structfield>dma_area</structfield> is necessary when the + buffer is mmapped. If your driver doesn't support mmap, this + field is not necessary. <structfield>dma_addr</structfield> + is also optional. You can use + <structfield>dma_private</structfield> as you like, too. + </para> + </section> + + <section id="pcm-interface-runtime-status"> + <title>Running Status</title> + <para> + The running status can be referred via <constant>runtime->status</constant>. + This is the pointer to the struct <structname>snd_pcm_mmap_status</structname> + record. For example, you can get the current DMA hardware + pointer via <constant>runtime->status->hw_ptr</constant>. + </para> + + <para> + The DMA application pointer can be referred via + <constant>runtime->control</constant>, which points to the + struct <structname>snd_pcm_mmap_control</structname> record. + However, accessing directly to this value is not recommended. + </para> + </section> + + <section id="pcm-interface-runtime-private"> + <title>Private Data</title> + <para> + You can allocate a record for the substream and store it in + <constant>runtime->private_data</constant>. Usually, this + is done in + <link linkend="pcm-interface-operators-open-callback"><citetitle> + the open callback</citetitle></link>. + Don't mix this with <constant>pcm->private_data</constant>. + The <constant>pcm->private_data</constant> usually points to the + chip instance assigned statically at the creation of PCM, while the + <constant>runtime->private_data</constant> points to a dynamic + data structure created at the PCM open callback. + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct my_pcm_data *data; + .... + data = kmalloc(sizeof(*data), GFP_KERNEL); + substream->runtime->private_data = data; + .... + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + The allocated object must be released in + <link linkend="pcm-interface-operators-open-callback"><citetitle> + the close callback</citetitle></link>. + </para> + </section> + + <section id="pcm-interface-runtime-intr"> + <title>Interrupt Callbacks</title> + <para> + The field <structfield>transfer_ack_begin</structfield> and + <structfield>transfer_ack_end</structfield> are called at + the beginning and at the end of + <function>snd_pcm_period_elapsed()</function>, respectively. + </para> + </section> + + </section> + + <section id="pcm-interface-operators"> + <title>Operators</title> + <para> + OK, now let me give details about each pcm callback + (<parameter>ops</parameter>). In general, every callback must + return 0 if successful, or a negative error number + such as <constant>-EINVAL</constant>. To choose an appropriate + error number, it is advised to check what value other parts of + the kernel return when the same kind of request fails. + </para> + + <para> + The callback function takes at least the argument with + <structname>snd_pcm_substream</structname> pointer. To retrieve + the chip record from the given substream instance, you can use the + following macro. + + <informalexample> + <programlisting> +<![CDATA[ + int xxx() { + struct mychip *chip = snd_pcm_substream_chip(substream); + .... + } +]]> + </programlisting> + </informalexample> + + The macro reads <constant>substream->private_data</constant>, + which is a copy of <constant>pcm->private_data</constant>. + You can override the former if you need to assign different data + records per PCM substream. For example, the cmi8330 driver assigns + different private_data for playback and capture directions, + because it uses two different codecs (SB- and AD-compatible) for + different directions. + </para> + + <section id="pcm-interface-operators-open-callback"> + <title>open callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_open(struct snd_pcm_substream *substream); +]]> + </programlisting> + </informalexample> + + This is called when a pcm substream is opened. + </para> + + <para> + At least, here you have to initialize the runtime->hw + record. Typically, this is done by like this: + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } +]]> + </programlisting> + </informalexample> + + where <parameter>snd_mychip_playback_hw</parameter> is the + pre-defined hardware description. + </para> + + <para> + You can allocate a private data in this callback, as described + in <link linkend="pcm-interface-runtime-private"><citetitle> + Private Data</citetitle></link> section. + </para> + + <para> + If the hardware configuration needs more constraints, set the + hardware constraints here, too. + See <link linkend="pcm-interface-constraints"><citetitle> + Constraints</citetitle></link> for more details. + </para> + </section> + + <section id="pcm-interface-operators-close-callback"> + <title>close callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_close(struct snd_pcm_substream *substream); +]]> + </programlisting> + </informalexample> + + Obviously, this is called when a pcm substream is closed. + </para> + + <para> + Any private instance for a pcm substream allocated in the + open callback will be released here. + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_close(struct snd_pcm_substream *substream) + { + .... + kfree(substream->runtime->private_data); + .... + } +]]> + </programlisting> + </informalexample> + </para> + </section> + + <section id="pcm-interface-operators-ioctl-callback"> + <title>ioctl callback</title> + <para> + This is used for any special call to pcm ioctls. But + usually you can pass a generic ioctl callback, + <function>snd_pcm_lib_ioctl</function>. + </para> + </section> + + <section id="pcm-interface-operators-hw-params-callback"> + <title>hw_params callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +]]> + </programlisting> + </informalexample> + </para> + + <para> + This is called when the hardware parameter + (<structfield>hw_params</structfield>) is set + up by the application, + that is, once when the buffer size, the period size, the + format, etc. are defined for the pcm substream. + </para> + + <para> + Many hardware setups should be done in this callback, + including the allocation of buffers. + </para> + + <para> + Parameters to be initialized are retrieved by + <function>params_xxx()</function> macros. To allocate + buffer, you can call a helper function, + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +]]> + </programlisting> + </informalexample> + + <function>snd_pcm_lib_malloc_pages()</function> is available + only when the DMA buffers have been pre-allocated. + See the section <link + linkend="buffer-and-memory-buffer-types"><citetitle> + Buffer Types</citetitle></link> for more details. + </para> + + <para> + Note that this and <structfield>prepare</structfield> callbacks + may be called multiple times per initialization. + For example, the OSS emulation may + call these callbacks at each change via its ioctl. + </para> + + <para> + Thus, you need to be careful not to allocate the same buffers + many times, which will lead to memory leaks! Calling the + helper function above many times is OK. It will release the + previous buffer automatically when it was already allocated. + </para> + + <para> + Another note is that this callback is non-atomic + (schedulable). This is important, because the + <structfield>trigger</structfield> callback + is atomic (non-schedulable). That is, mutexes or any + schedule-related functions are not available in + <structfield>trigger</structfield> callback. + Please see the subsection + <link linkend="pcm-interface-atomicity"><citetitle> + Atomicity</citetitle></link> for details. + </para> + </section> + + <section id="pcm-interface-operators-hw-free-callback"> + <title>hw_free callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_hw_free(struct snd_pcm_substream *substream); +]]> + </programlisting> + </informalexample> + </para> + + <para> + This is called to release the resources allocated via + <structfield>hw_params</structfield>. For example, releasing the + buffer via + <function>snd_pcm_lib_malloc_pages()</function> is done by + calling the following: + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_lib_free_pages(substream); +]]> + </programlisting> + </informalexample> + </para> + + <para> + This function is always called before the close callback is called. + Also, the callback may be called multiple times, too. + Keep track whether the resource was already released. + </para> + </section> + + <section id="pcm-interface-operators-prepare-callback"> + <title>prepare callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_prepare(struct snd_pcm_substream *substream); +]]> + </programlisting> + </informalexample> + </para> + + <para> + This callback is called when the pcm is + <quote>prepared</quote>. You can set the format type, sample + rate, etc. here. The difference from + <structfield>hw_params</structfield> is that the + <structfield>prepare</structfield> callback will be called each + time + <function>snd_pcm_prepare()</function> is called, i.e. when + recovering after underruns, etc. + </para> + + <para> + Note that this callback is now non-atomic. + You can use schedule-related functions safely in this callback. + </para> + + <para> + In this and the following callbacks, you can refer to the + values via the runtime record, + substream->runtime. + For example, to get the current + rate, format or channels, access to + runtime->rate, + runtime->format or + runtime->channels, respectively. + The physical address of the allocated buffer is set to + runtime->dma_area. The buffer and period sizes are + in runtime->buffer_size and runtime->period_size, + respectively. + </para> + + <para> + Be careful that this callback will be called many times at + each setup, too. + </para> + </section> + + <section id="pcm-interface-operators-trigger-callback"> + <title>trigger callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); +]]> + </programlisting> + </informalexample> + + This is called when the pcm is started, stopped or paused. + </para> + + <para> + Which action is specified in the second argument, + <constant>SNDRV_PCM_TRIGGER_XXX</constant> in + <filename><sound/pcm.h></filename>. At least, + the <constant>START</constant> and <constant>STOP</constant> + commands must be defined in this callback. + + <informalexample> + <programlisting> +<![CDATA[ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + break; + default: + return -EINVAL; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + When the pcm supports the pause operation (given in the info + field of the hardware table), the <constant>PAUSE_PUSE</constant> + and <constant>PAUSE_RELEASE</constant> commands must be + handled here, too. The former is the command to pause the pcm, + and the latter to restart the pcm again. + </para> + + <para> + When the pcm supports the suspend/resume operation, + regardless of full or partial suspend/resume support, + the <constant>SUSPEND</constant> and <constant>RESUME</constant> + commands must be handled, too. + These commands are issued when the power-management status is + changed. Obviously, the <constant>SUSPEND</constant> and + <constant>RESUME</constant> commands + suspend and resume the pcm substream, and usually, they + are identical to the <constant>STOP</constant> and + <constant>START</constant> commands, respectively. + See the <link linkend="power-management"><citetitle> + Power Management</citetitle></link> section for details. + </para> + + <para> + As mentioned, this callback is atomic. You cannot call + functions which may sleep. + The trigger callback should be as minimal as possible, + just really triggering the DMA. The other stuff should be + initialized hw_params and prepare callbacks properly + beforehand. + </para> + </section> + + <section id="pcm-interface-operators-pointer-callback"> + <title>pointer callback</title> + <para> + <informalexample> + <programlisting> +<![CDATA[ + static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) +]]> + </programlisting> + </informalexample> + + This callback is called when the PCM middle layer inquires + the current hardware position on the buffer. The position must + be returned in frames, + ranging from 0 to buffer_size - 1. + </para> + + <para> + This is called usually from the buffer-update routine in the + pcm middle layer, which is invoked when + <function>snd_pcm_period_elapsed()</function> is called in the + interrupt routine. Then the pcm middle layer updates the + position and calculates the available space, and wakes up the + sleeping poll threads, etc. + </para> + + <para> + This callback is also atomic. + </para> + </section> + + <section id="pcm-interface-operators-copy-silence"> + <title>copy and silence callbacks</title> + <para> + These callbacks are not mandatory, and can be omitted in + most cases. These callbacks are used when the hardware buffer + cannot be in the normal memory space. Some chips have their + own buffer on the hardware which is not mappable. In such a + case, you have to transfer the data manually from the memory + buffer to the hardware buffer. Or, if the buffer is + non-contiguous on both physical and virtual memory spaces, + these callbacks must be defined, too. + </para> + + <para> + If these two callbacks are defined, copy and set-silence + operations are done by them. The detailed will be described in + the later section <link + linkend="buffer-and-memory"><citetitle>Buffer and Memory + Management</citetitle></link>. + </para> + </section> + + <section id="pcm-interface-operators-ack"> + <title>ack callback</title> + <para> + This callback is also not mandatory. This callback is called + when the appl_ptr is updated in read or write operations. + Some drivers like emu10k1-fx and cs46xx need to track the + current appl_ptr for the internal buffer, and this callback + is useful only for such a purpose. + </para> + <para> + This callback is atomic. + </para> + </section> + + <section id="pcm-interface-operators-page-callback"> + <title>page callback</title> + + <para> + This callback is optional too. This callback is used + mainly for non-contiguous buffers. The mmap calls this + callback to get the page address. Some examples will be + explained in the later section <link + linkend="buffer-and-memory"><citetitle>Buffer and Memory + Management</citetitle></link>, too. + </para> + </section> + </section> + + <section id="pcm-interface-interrupt-handler"> + <title>Interrupt Handler</title> + <para> + The rest of pcm stuff is the PCM interrupt handler. The + role of PCM interrupt handler in the sound driver is to update + the buffer position and to tell the PCM middle layer when the + buffer position goes across the prescribed period size. To + inform this, call the <function>snd_pcm_period_elapsed()</function> + function. + </para> + + <para> + There are several types of sound chips to generate the interrupts. + </para> + + <section id="pcm-interface-interrupt-handler-boundary"> + <title>Interrupts at the period (fragment) boundary</title> + <para> + This is the most frequently found type: the hardware + generates an interrupt at each period boundary. + In this case, you can call + <function>snd_pcm_period_elapsed()</function> at each + interrupt. + </para> + + <para> + <function>snd_pcm_period_elapsed()</function> takes the + substream pointer as its argument. Thus, you need to keep the + substream pointer accessible from the chip instance. For + example, define substream field in the chip record to hold the + current running substream pointer, and set the pointer value + at open callback (and reset at close callback). + </para> + + <para> + If you acquire a spinlock in the interrupt handler, and the + lock is used in other pcm callbacks, too, then you have to + release the lock before calling + <function>snd_pcm_period_elapsed()</function>, because + <function>snd_pcm_period_elapsed()</function> calls other pcm + callbacks inside. + </para> + + <para> + Typical code would be like: + + <example> + <title>Interrupt Handler Case #1</title> + <programlisting> +<![CDATA[ + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + /* call updater, unlock before it */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(chip->substream); + spin_lock(&chip->lock); + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } +]]> + </programlisting> + </example> + </para> + </section> + + <section id="pcm-interface-interrupt-handler-timer"> + <title>High frequency timer interrupts</title> + <para> + This happense when the hardware doesn't generate interrupts + at the period boundary but issues timer interrupts at a fixed + timer rate (e.g. es1968 or ymfpci drivers). + In this case, you need to check the current hardware + position and accumulate the processed sample length at each + interrupt. When the accumulated size exceeds the period + size, call + <function>snd_pcm_period_elapsed()</function> and reset the + accumulator. + </para> + + <para> + Typical code would be like the following. + + <example> + <title>Interrupt Handler Case #2</title> + <programlisting> +<![CDATA[ + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + unsigned int last_ptr, size; + /* get the current hardware pointer (in frames) */ + last_ptr = get_hw_ptr(chip); + /* calculate the processed frames since the + * last update + */ + if (last_ptr < chip->last_ptr) + size = runtime->buffer_size + last_ptr + - chip->last_ptr; + else + size = last_ptr - chip->last_ptr; + /* remember the last updated point */ + chip->last_ptr = last_ptr; + /* accumulate the size */ + chip->size += size; + /* over the period boundary? */ + if (chip->size >= runtime->period_size) { + /* reset the accumulator */ + chip->size %= runtime->period_size; + /* call updater */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(substream); + spin_lock(&chip->lock); + } + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } +]]> + </programlisting> + </example> + </para> + </section> + + <section id="pcm-interface-interrupt-handler-both"> + <title>On calling <function>snd_pcm_period_elapsed()</function></title> + <para> + In both cases, even if more than one period are elapsed, you + don't have to call + <function>snd_pcm_period_elapsed()</function> many times. Call + only once. And the pcm layer will check the current hardware + pointer and update to the latest status. + </para> + </section> + </section> + + <section id="pcm-interface-atomicity"> + <title>Atomicity</title> + <para> + One of the most important (and thus difficult to debug) problems + in kernel programming are race conditions. + In the Linux kernel, they are usually avoided via spin-locks, mutexes + or semaphores. In general, if a race condition can happen + in an interrupt handler, it has to be managed atomically, and you + have to use a spinlock to protect the critical session. If the + critical section is not in interrupt handler code and + if taking a relatively long time to execute is acceptable, you + should use mutexes or semaphores instead. + </para> + + <para> + As already seen, some pcm callbacks are atomic and some are + not. For example, the <parameter>hw_params</parameter> callback is + non-atomic, while <parameter>trigger</parameter> callback is + atomic. This means, the latter is called already in a spinlock + held by the PCM middle layer. Please take this atomicity into + account when you choose a locking scheme in the callbacks. + </para> + + <para> + In the atomic callbacks, you cannot use functions which may call + <function>schedule</function> or go to + <function>sleep</function>. Semaphores and mutexes can sleep, + and hence they cannot be used inside the atomic callbacks + (e.g. <parameter>trigger</parameter> callback). + To implement some delay in such a callback, please use + <function>udelay()</function> or <function>mdelay()</function>. + </para> + + <para> + All three atomic callbacks (trigger, pointer, and ack) are + called with local interrupts disabled. + </para> + + </section> + <section id="pcm-interface-constraints"> + <title>Constraints</title> + <para> + If your chip supports unconventional sample rates, or only the + limited samples, you need to set a constraint for the + condition. + </para> + + <para> + For example, in order to restrict the sample rates in the some + supported values, use + <function>snd_pcm_hw_constraint_list()</function>. + You need to call this function in the open callback. + + <example> + <title>Example of Hardware Constraints</title> + <programlisting> +<![CDATA[ + static unsigned int rates[] = + {4000, 10000, 22050, 44100}; + static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + static int snd_mychip_pcm_open(struct snd_pcm_substream *substream) + { + int err; + .... + err = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } +]]> + </programlisting> + </example> + </para> + + <para> + There are many different constraints. + Look at <filename>sound/pcm.h</filename> for a complete list. + You can even define your own constraint rules. + For example, let's suppose my_chip can manage a substream of 1 channel + if and only if the format is S16_LE, otherwise it supports any format + specified in the <structname>snd_pcm_hardware</structname> structure (or in any + other constraint_list). You can build a rule like this: + + <example> + <title>Example of Hardware Constraints for Channels</title> + <programlisting> +<![CDATA[ + static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask fmt; + + snd_mask_any(&fmt); /* Init the struct */ + if (c->min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } +]]> + </programlisting> + </example> + </para> + + <para> + Then you need to call this function to add your rule: + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, + -1); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The rule function is called when an application sets the number of + channels. But an application can set the format before the number of + channels. Thus you also need to define the inverse rule: + + <example> + <title>Example of Hardware Constraints for Channels</title> + <programlisting> +<![CDATA[ + static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval ch; + + snd_interval_any(&ch); + if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } +]]> + </programlisting> + </example> + </para> + + <para> + ...and in the open callback: + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); +]]> + </programlisting> + </informalexample> + </para> + + <para> + I won't give more details here, rather I + would like to say, <quote>Luke, use the source.</quote> + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Control Interface --> +<!-- ****************************************************** --> + <chapter id="control-interface"> + <title>Control Interface</title> + + <section id="control-interface-general"> + <title>General</title> + <para> + The control interface is used widely for many switches, + sliders, etc. which are accessed from user-space. Its most + important use is the mixer interface. In other words, since ALSA + 0.9.x, all the mixer stuff is implemented on the control kernel API. + </para> + + <para> + ALSA has a well-defined AC97 control module. If your chip + supports only the AC97 and nothing else, you can skip this + section. + </para> + + <para> + The control API is defined in + <filename><sound/control.h></filename>. + Include this file if you want to add your own controls. + </para> + </section> + + <section id="control-interface-definition"> + <title>Definition of Controls</title> + <para> + To create a new control, you need to define the + following three + callbacks: <structfield>info</structfield>, + <structfield>get</structfield> and + <structfield>put</structfield>. Then, define a + struct <structname>snd_kcontrol_new</structname> record, such as: + + <example> + <title>Definition of a Control</title> + <programlisting> +<![CDATA[ + static struct snd_kcontrol_new my_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = my_control_info, + .get = my_control_get, + .put = my_control_put + }; +]]> + </programlisting> + </example> + </para> + + <para> + Most likely the control is created via + <function>snd_ctl_new1()</function>, and in such a case, you can + add the <parameter>__devinitdata</parameter> prefix to the + definition as above. + </para> + + <para> + The <structfield>iface</structfield> field specifies the control + type, <constant>SNDRV_CTL_ELEM_IFACE_XXX</constant>, which + is usually <constant>MIXER</constant>. + Use <constant>CARD</constant> for global controls that are not + logically part of the mixer. + If the control is closely associated with some specific device on + the sound card, use <constant>HWDEP</constant>, + <constant>PCM</constant>, <constant>RAWMIDI</constant>, + <constant>TIMER</constant>, or <constant>SEQUENCER</constant>, and + specify the device number with the + <structfield>device</structfield> and + <structfield>subdevice</structfield> fields. + </para> + + <para> + The <structfield>name</structfield> is the name identifier + string. Since ALSA 0.9.x, the control name is very important, + because its role is classified from its name. There are + pre-defined standard control names. The details are described in + the <link linkend="control-interface-control-names"><citetitle> + Control Names</citetitle></link> subsection. + </para> + + <para> + The <structfield>index</structfield> field holds the index number + of this control. If there are several different controls with + the same name, they can be distinguished by the index + number. This is the case when + several codecs exist on the card. If the index is zero, you can + omit the definition above. + </para> + + <para> + The <structfield>access</structfield> field contains the access + type of this control. Give the combination of bit masks, + <constant>SNDRV_CTL_ELEM_ACCESS_XXX</constant>, there. + The details will be explained in + the <link linkend="control-interface-access-flags"><citetitle> + Access Flags</citetitle></link> subsection. + </para> + + <para> + The <structfield>private_value</structfield> field contains + an arbitrary long integer value for this record. When using + the generic <structfield>info</structfield>, + <structfield>get</structfield> and + <structfield>put</structfield> callbacks, you can pass a value + through this field. If several small numbers are necessary, you can + combine them in bitwise. Or, it's possible to give a pointer + (casted to unsigned long) of some record to this field, too. + </para> + + <para> + The <structfield>tlv</structfield> field can be used to provide + metadata about the control; see the + <link linkend="control-interface-tlv"> + <citetitle>Metadata</citetitle></link> subsection. + </para> + + <para> + The other three are + <link linkend="control-interface-callbacks"><citetitle> + callback functions</citetitle></link>. + </para> + </section> + + <section id="control-interface-control-names"> + <title>Control Names</title> + <para> + There are some standards to define the control names. A + control is usually defined from the three parts as + <quote>SOURCE DIRECTION FUNCTION</quote>. + </para> + + <para> + The first, <constant>SOURCE</constant>, specifies the source + of the control, and is a string such as <quote>Master</quote>, + <quote>PCM</quote>, <quote>CD</quote> and + <quote>Line</quote>. There are many pre-defined sources. + </para> + + <para> + The second, <constant>DIRECTION</constant>, is one of the + following strings according to the direction of the control: + <quote>Playback</quote>, <quote>Capture</quote>, <quote>Bypass + Playback</quote> and <quote>Bypass Capture</quote>. Or, it can + be omitted, meaning both playback and capture directions. + </para> + + <para> + The third, <constant>FUNCTION</constant>, is one of the + following strings according to the function of the control: + <quote>Switch</quote>, <quote>Volume</quote> and + <quote>Route</quote>. + </para> + + <para> + The example of control names are, thus, <quote>Master Capture + Switch</quote> or <quote>PCM Playback Volume</quote>. + </para> + + <para> + There are some exceptions: + </para> + + <section id="control-interface-control-names-global"> + <title>Global capture and playback</title> + <para> + <quote>Capture Source</quote>, <quote>Capture Switch</quote> + and <quote>Capture Volume</quote> are used for the global + capture (input) source, switch and volume. Similarly, + <quote>Playback Switch</quote> and <quote>Playback + Volume</quote> are used for the global output gain switch and + volume. + </para> + </section> + + <section id="control-interface-control-names-tone"> + <title>Tone-controls</title> + <para> + tone-control switch and volumes are specified like + <quote>Tone Control - XXX</quote>, e.g. <quote>Tone Control - + Switch</quote>, <quote>Tone Control - Bass</quote>, + <quote>Tone Control - Center</quote>. + </para> + </section> + + <section id="control-interface-control-names-3d"> + <title>3D controls</title> + <para> + 3D-control switches and volumes are specified like <quote>3D + Control - XXX</quote>, e.g. <quote>3D Control - + Switch</quote>, <quote>3D Control - Center</quote>, <quote>3D + Control - Space</quote>. + </para> + </section> + + <section id="control-interface-control-names-mic"> + <title>Mic boost</title> + <para> + Mic-boost switch is set as <quote>Mic Boost</quote> or + <quote>Mic Boost (6dB)</quote>. + </para> + + <para> + More precise information can be found in + <filename>Documentation/sound/alsa/ControlNames.txt</filename>. + </para> + </section> + </section> + + <section id="control-interface-access-flags"> + <title>Access Flags</title> + + <para> + The access flag is the bitmask which specifies the access type + of the given control. The default access type is + <constant>SNDRV_CTL_ELEM_ACCESS_READWRITE</constant>, + which means both read and write are allowed to this control. + When the access flag is omitted (i.e. = 0), it is + considered as <constant>READWRITE</constant> access as default. + </para> + + <para> + When the control is read-only, pass + <constant>SNDRV_CTL_ELEM_ACCESS_READ</constant> instead. + In this case, you don't have to define + the <structfield>put</structfield> callback. + Similarly, when the control is write-only (although it's a rare + case), you can use the <constant>WRITE</constant> flag instead, and + you don't need the <structfield>get</structfield> callback. + </para> + + <para> + If the control value changes frequently (e.g. the VU meter), + <constant>VOLATILE</constant> flag should be given. This means + that the control may be changed without + <link linkend="control-interface-change-notification"><citetitle> + notification</citetitle></link>. Applications should poll such + a control constantly. + </para> + + <para> + When the control is inactive, set + the <constant>INACTIVE</constant> flag, too. + There are <constant>LOCK</constant> and + <constant>OWNER</constant> flags to change the write + permissions. + </para> + + </section> + + <section id="control-interface-callbacks"> + <title>Callbacks</title> + + <section id="control-interface-callbacks-info"> + <title>info callback</title> + <para> + The <structfield>info</structfield> callback is used to get + detailed information on this control. This must store the + values of the given struct <structname>snd_ctl_elem_info</structname> + object. For example, for a boolean control with a single + element: + + <example> + <title>Example of info callback</title> + <programlisting> +<![CDATA[ + static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } +]]> + </programlisting> + </example> + </para> + + <para> + The <structfield>type</structfield> field specifies the type + of the control. There are <constant>BOOLEAN</constant>, + <constant>INTEGER</constant>, <constant>ENUMERATED</constant>, + <constant>BYTES</constant>, <constant>IEC958</constant> and + <constant>INTEGER64</constant>. The + <structfield>count</structfield> field specifies the + number of elements in this control. For example, a stereo + volume would have count = 2. The + <structfield>value</structfield> field is a union, and + the values stored are depending on the type. The boolean and + integer types are identical. + </para> + + <para> + The enumerated type is a bit different from others. You'll + need to set the string for the currently given item index. + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + Some common info callbacks are available for your convenience: + <function>snd_ctl_boolean_mono_info()</function> and + <function>snd_ctl_boolean_stereo_info()</function>. + Obviously, the former is an info callback for a mono channel + boolean item, just like <function>snd_myctl_mono_info</function> + above, and the latter is for a stereo channel boolean item. + </para> + + </section> + + <section id="control-interface-callbacks-get"> + <title>get callback</title> + + <para> + This callback is used to read the current value of the + control and to return to user-space. + </para> + + <para> + For example, + + <example> + <title>Example of get callback</title> + <programlisting> +<![CDATA[ + static int snd_myctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = get_some_value(chip); + return 0; + } +]]> + </programlisting> + </example> + </para> + + <para> + The <structfield>value</structfield> field depends on + the type of control as well as on the info callback. For example, + the sb driver uses this field to store the register offset, + the bit-shift and the bit-mask. The + <structfield>private_value</structfield> field is set as follows: + <informalexample> + <programlisting> +<![CDATA[ + .private_value = reg | (shift << 16) | (mask << 24) +]]> + </programlisting> + </informalexample> + and is retrieved in callbacks like + <informalexample> + <programlisting> +<![CDATA[ + static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + In the <structfield>get</structfield> callback, + you have to fill all the elements if the + control has more than one elements, + i.e. <structfield>count</structfield> > 1. + In the example above, we filled only one element + (<structfield>value.integer.value[0]</structfield>) since it's + assumed as <structfield>count</structfield> = 1. + </para> + </section> + + <section id="control-interface-callbacks-put"> + <title>put callback</title> + + <para> + This callback is used to write a value from user-space. + </para> + + <para> + For example, + + <example> + <title>Example of put callback</title> + <programlisting> +<![CDATA[ + static int snd_myctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (chip->current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } +]]> + </programlisting> + </example> + + As seen above, you have to return 1 if the value is + changed. If the value is not changed, return 0 instead. + If any fatal error happens, return a negative error code as + usual. + </para> + + <para> + As in the <structfield>get</structfield> callback, + when the control has more than one elements, + all elements must be evaluated in this callback, too. + </para> + </section> + + <section id="control-interface-callbacks-all"> + <title>Callbacks are not atomic</title> + <para> + All these three callbacks are basically not atomic. + </para> + </section> + </section> + + <section id="control-interface-constructor"> + <title>Constructor</title> + <para> + When everything is ready, finally we can create a new + control. To create a control, there are two functions to be + called, <function>snd_ctl_new1()</function> and + <function>snd_ctl_add()</function>. + </para> + + <para> + In the simplest way, you can do like this: + + <informalexample> + <programlisting> +<![CDATA[ + err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); + if (err < 0) + return err; +]]> + </programlisting> + </informalexample> + + where <parameter>my_control</parameter> is the + struct <structname>snd_kcontrol_new</structname> object defined above, and chip + is the object pointer to be passed to + kcontrol->private_data + which can be referred to in callbacks. + </para> + + <para> + <function>snd_ctl_new1()</function> allocates a new + <structname>snd_kcontrol</structname> instance (that's why the definition + of <parameter>my_control</parameter> can be with + the <parameter>__devinitdata</parameter> + prefix), and <function>snd_ctl_add</function> assigns the given + control component to the card. + </para> + </section> + + <section id="control-interface-change-notification"> + <title>Change Notification</title> + <para> + If you need to change and update a control in the interrupt + routine, you can call <function>snd_ctl_notify()</function>. For + example, + + <informalexample> + <programlisting> +<![CDATA[ + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); +]]> + </programlisting> + </informalexample> + + This function takes the card pointer, the event-mask, and the + control id pointer for the notification. The event-mask + specifies the types of notification, for example, in the above + example, the change of control values is notified. + The id pointer is the pointer of struct <structname>snd_ctl_elem_id</structname> + to be notified. + You can find some examples in <filename>es1938.c</filename> or + <filename>es1968.c</filename> for hardware volume interrupts. + </para> + </section> + + <section id="control-interface-tlv"> + <title>Metadata</title> + <para> + To provide information about the dB values of a mixer control, use + on of the <constant>DECLARE_TLV_xxx</constant> macros from + <filename><sound/tlv.h></filename> to define a variable + containing this information, set the<structfield>tlv.p + </structfield> field to point to this variable, and include the + <constant>SNDRV_CTL_ELEM_ACCESS_TLV_READ</constant> flag in the + <structfield>access</structfield> field; like this: + <informalexample> + <programlisting> +<![CDATA[ + static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); + + static struct snd_kcontrol_new my_control __devinitdata = { + ... + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + ... + .tlv.p = db_scale_my_control, + }; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The <function>DECLARE_TLV_DB_SCALE</function> macro defines + information about a mixer control where each step in the control's + value changes the dB value by a constant dB amount. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the step size, in units of 0.01 dB. + Set the fourth parameter to 1 if the minimum value actually mutes + the control. + </para> + + <para> + The <function>DECLARE_TLV_DB_LINEAR</function> macro defines + information about a mixer control where the control's value affects + the output linearly. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the maximum value, in units of 0.01 dB. + If the minimum value mutes the control, set the second parameter to + <constant>TLV_DB_GAIN_MUTE</constant>. + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- API for AC97 Codec --> +<!-- ****************************************************** --> + <chapter id="api-ac97"> + <title>API for AC97 Codec</title> + + <section> + <title>General</title> + <para> + The ALSA AC97 codec layer is a well-defined one, and you don't + have to write much code to control it. Only low-level control + routines are necessary. The AC97 codec API is defined in + <filename><sound/ac97_codec.h></filename>. + </para> + </section> + + <section id="api-ac97-example"> + <title>Full Code Example</title> + <para> + <example> + <title>Example of AC97 Interface</title> + <programlisting> +<![CDATA[ + struct mychip { + .... + struct snd_ac97 *ac97; + .... + }; + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + /* read a register value here from the codec */ + return the_register_value; + } + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + { + struct mychip *chip = ac97->private_data; + .... + /* write the given register value to the codec */ + } + + static int snd_mychip_ac97(struct mychip *chip) + { + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + +]]> + </programlisting> + </example> + </para> + </section> + + <section id="api-ac97-constructor"> + <title>Constructor</title> + <para> + To create an ac97 instance, first call <function>snd_ac97_bus</function> + with an <type>ac97_bus_ops_t</type> record with callback functions. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_ac97_bus *bus; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + snd_ac97_bus(card, 0, &ops, NULL, &pbus); +]]> + </programlisting> + </informalexample> + + The bus record is shared among all belonging ac97 instances. + </para> + + <para> + And then call <function>snd_ac97_mixer()</function> with an + struct <structname>snd_ac97_template</structname> + record together with the bus pointer created above. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_ac97_template ac97; + int err; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + snd_ac97_mixer(bus, &ac97, &chip->ac97); +]]> + </programlisting> + </informalexample> + + where chip->ac97 is a pointer to a newly created + <type>ac97_t</type> instance. + In this case, the chip pointer is set as the private data, so that + the read/write callback functions can refer to this chip instance. + This instance is not necessarily stored in the chip + record. If you need to change the register values from the + driver, or need the suspend/resume of ac97 codecs, keep this + pointer to pass to the corresponding functions. + </para> + </section> + + <section id="api-ac97-callbacks"> + <title>Callbacks</title> + <para> + The standard callbacks are <structfield>read</structfield> and + <structfield>write</structfield>. Obviously they + correspond to the functions for read and write accesses to the + hardware low-level codes. + </para> + + <para> + The <structfield>read</structfield> callback returns the + register value specified in the argument. + + <informalexample> + <programlisting> +<![CDATA[ + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + return the_register_value; + } +]]> + </programlisting> + </informalexample> + + Here, the chip can be cast from ac97->private_data. + </para> + + <para> + Meanwhile, the <structfield>write</structfield> callback is + used to set the register value. + + <informalexample> + <programlisting> +<![CDATA[ + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +]]> + </programlisting> + </informalexample> + </para> + + <para> + These callbacks are non-atomic like the control API callbacks. + </para> + + <para> + There are also other callbacks: + <structfield>reset</structfield>, + <structfield>wait</structfield> and + <structfield>init</structfield>. + </para> + + <para> + The <structfield>reset</structfield> callback is used to reset + the codec. If the chip requires a special kind of reset, you can + define this callback. + </para> + + <para> + The <structfield>wait</structfield> callback is used to + add some waiting time in the standard initialization of the codec. If the + chip requires the extra waiting time, define this callback. + </para> + + <para> + The <structfield>init</structfield> callback is used for + additional initialization of the codec. + </para> + </section> + + <section id="api-ac97-updating-registers"> + <title>Updating Registers in The Driver</title> + <para> + If you need to access to the codec from the driver, you can + call the following functions: + <function>snd_ac97_write()</function>, + <function>snd_ac97_read()</function>, + <function>snd_ac97_update()</function> and + <function>snd_ac97_update_bits()</function>. + </para> + + <para> + Both <function>snd_ac97_write()</function> and + <function>snd_ac97_update()</function> functions are used to + set a value to the given register + (<constant>AC97_XXX</constant>). The difference between them is + that <function>snd_ac97_update()</function> doesn't write a + value if the given value has been already set, while + <function>snd_ac97_write()</function> always rewrites the + value. + + <informalexample> + <programlisting> +<![CDATA[ + snd_ac97_write(ac97, AC97_MASTER, 0x8080); + snd_ac97_update(ac97, AC97_MASTER, 0x8080); +]]> + </programlisting> + </informalexample> + </para> + + <para> + <function>snd_ac97_read()</function> is used to read the value + of the given register. For example, + + <informalexample> + <programlisting> +<![CDATA[ + value = snd_ac97_read(ac97, AC97_MASTER); +]]> + </programlisting> + </informalexample> + </para> + + <para> + <function>snd_ac97_update_bits()</function> is used to update + some bits in the given register. + + <informalexample> + <programlisting> +<![CDATA[ + snd_ac97_update_bits(ac97, reg, mask, value); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Also, there is a function to change the sample rate (of a + given register such as + <constant>AC97_PCM_FRONT_DAC_RATE</constant>) when VRA or + DRA is supported by the codec: + <function>snd_ac97_set_rate()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The following registers are available to set the rate: + <constant>AC97_PCM_MIC_ADC_RATE</constant>, + <constant>AC97_PCM_FRONT_DAC_RATE</constant>, + <constant>AC97_PCM_LR_ADC_RATE</constant>, + <constant>AC97_SPDIF</constant>. When + <constant>AC97_SPDIF</constant> is specified, the register is + not really changed but the corresponding IEC958 status bits will + be updated. + </para> + </section> + + <section id="api-ac97-clock-adjustment"> + <title>Clock Adjustment</title> + <para> + In some chips, the clock of the codec isn't 48000 but using a + PCI clock (to save a quartz!). In this case, change the field + bus->clock to the corresponding + value. For example, intel8x0 + and es1968 drivers have their own function to read from the clock. + </para> + </section> + + <section id="api-ac97-proc-files"> + <title>Proc Files</title> + <para> + The ALSA AC97 interface will create a proc file such as + <filename>/proc/asound/card0/codec97#0/ac97#0-0</filename> and + <filename>ac97#0-0+regs</filename>. You can refer to these files to + see the current status and registers of the codec. + </para> + </section> + + <section id="api-ac97-multiple-codecs"> + <title>Multiple Codecs</title> + <para> + When there are several codecs on the same card, you need to + call <function>snd_ac97_mixer()</function> multiple times with + ac97.num=1 or greater. The <structfield>num</structfield> field + specifies the codec number. + </para> + + <para> + If you set up multiple codecs, you either need to write + different callbacks for each codec or check + ac97->num in the callback routines. + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- MIDI (MPU401-UART) Interface --> +<!-- ****************************************************** --> + <chapter id="midi-interface"> + <title>MIDI (MPU401-UART) Interface</title> + + <section id="midi-interface-general"> + <title>General</title> + <para> + Many soundcards have built-in MIDI (MPU401-UART) + interfaces. When the soundcard supports the standard MPU401-UART + interface, most likely you can use the ALSA MPU401-UART API. The + MPU401-UART API is defined in + <filename><sound/mpu401.h></filename>. + </para> + + <para> + Some soundchips have a similar but slightly different + implementation of mpu401 stuff. For example, emu10k1 has its own + mpu401 routines. + </para> + </section> + + <section id="midi-interface-constructor"> + <title>Constructor</title> + <para> + To create a rawmidi object, call + <function>snd_mpu401_uart_new()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_rawmidi *rmidi; + snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, + irq, irq_flags, &rmidi); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The first argument is the card pointer, and the second is the + index of this component. You can create up to 8 rawmidi + devices. + </para> + + <para> + The third argument is the type of the hardware, + <constant>MPU401_HW_XXX</constant>. If it's not a special one, + you can use <constant>MPU401_HW_MPU401</constant>. + </para> + + <para> + The 4th argument is the I/O port address. Many + backward-compatible MPU401 have an I/O port such as 0x330. Or, it + might be a part of its own PCI I/O region. It depends on the + chip design. + </para> + + <para> + The 5th argument is a bitflag for additional information. + When the I/O port address above is part of the PCI I/O + region, the MPU401 I/O port might have been already allocated + (reserved) by the driver itself. In such a case, pass a bit flag + <constant>MPU401_INFO_INTEGRATED</constant>, + and the mpu401-uart layer will allocate the I/O ports by itself. + </para> + + <para> + When the controller supports only the input or output MIDI stream, + pass the <constant>MPU401_INFO_INPUT</constant> or + <constant>MPU401_INFO_OUTPUT</constant> bitflag, respectively. + Then the rawmidi instance is created as a single stream. + </para> + + <para> + <constant>MPU401_INFO_MMIO</constant> bitflag is used to change + the access method to MMIO (via readb and writeb) instead of + iob and outb. In this case, you have to pass the iomapped address + to <function>snd_mpu401_uart_new()</function>. + </para> + + <para> + When <constant>MPU401_INFO_TX_IRQ</constant> is set, the output + stream isn't checked in the default interrupt handler. The driver + needs to call <function>snd_mpu401_uart_interrupt_tx()</function> + by itself to start processing the output stream in the irq handler. + </para> + + <para> + Usually, the port address corresponds to the command port and + port + 1 corresponds to the data port. If not, you may change + the <structfield>cport</structfield> field of + struct <structname>snd_mpu401</structname> manually + afterward. However, <structname>snd_mpu401</structname> pointer is not + returned explicitly by + <function>snd_mpu401_uart_new()</function>. You need to cast + rmidi->private_data to + <structname>snd_mpu401</structname> explicitly, + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_mpu401 *mpu; + mpu = rmidi->private_data; +]]> + </programlisting> + </informalexample> + + and reset the cport as you like: + + <informalexample> + <programlisting> +<![CDATA[ + mpu->cport = my_own_control_port; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The 6th argument specifies the irq number for UART. If the irq + is already allocated, pass 0 to the 7th argument + (<parameter>irq_flags</parameter>). Otherwise, pass the flags + for irq allocation + (<constant>SA_XXX</constant> bits) to it, and the irq will be + reserved by the mpu401-uart layer. If the card doesn't generate + UART interrupts, pass -1 as the irq number. Then a timer + interrupt will be invoked for polling. + </para> + </section> + + <section id="midi-interface-interrupt-handler"> + <title>Interrupt Handler</title> + <para> + When the interrupt is allocated in + <function>snd_mpu401_uart_new()</function>, the private + interrupt handler is used, hence you don't have anything else to do + than creating the mpu401 stuff. Otherwise, you have to call + <function>snd_mpu401_uart_interrupt()</function> explicitly when + a UART interrupt is invoked and checked in your own interrupt + handler. + </para> + + <para> + In this case, you need to pass the private_data of the + returned rawmidi object from + <function>snd_mpu401_uart_new()</function> as the second + argument of <function>snd_mpu401_uart_interrupt()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); +]]> + </programlisting> + </informalexample> + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- RawMIDI Interface --> +<!-- ****************************************************** --> + <chapter id="rawmidi-interface"> + <title>RawMIDI Interface</title> + + <section id="rawmidi-interface-overview"> + <title>Overview</title> + + <para> + The raw MIDI interface is used for hardware MIDI ports that can + be accessed as a byte stream. It is not used for synthesizer + chips that do not directly understand MIDI. + </para> + + <para> + ALSA handles file and buffer management. All you have to do is + to write some code to move data between the buffer and the + hardware. + </para> + + <para> + The rawmidi API is defined in + <filename><sound/rawmidi.h></filename>. + </para> + </section> + + <section id="rawmidi-interface-constructor"> + <title>Constructor</title> + + <para> + To create a rawmidi device, call the + <function>snd_rawmidi_new</function> function: + <informalexample> + <programlisting> +<![CDATA[ + struct snd_rawmidi *rmidi; + err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); + if (err < 0) + return err; + rmidi->private_data = chip; + strcpy(rmidi->name, "My MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The first argument is the card pointer, the second argument is + the ID string. + </para> + + <para> + The third argument is the index of this component. You can + create up to 8 rawmidi devices. + </para> + + <para> + The fourth and fifth arguments are the number of output and + input substreams, respectively, of this device (a substream is + the equivalent of a MIDI port). + </para> + + <para> + Set the <structfield>info_flags</structfield> field to specify + the capabilities of the device. + Set <constant>SNDRV_RAWMIDI_INFO_OUTPUT</constant> if there is + at least one output port, + <constant>SNDRV_RAWMIDI_INFO_INPUT</constant> if there is at + least one input port, + and <constant>SNDRV_RAWMIDI_INFO_DUPLEX</constant> if the device + can handle output and input at the same time. + </para> + + <para> + After the rawmidi device is created, you need to set the + operators (callbacks) for each substream. There are helper + functions to set the operators for all the substreams of a device: + <informalexample> + <programlisting> +<![CDATA[ + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The operators are usually defined like this: + <informalexample> + <programlisting> +<![CDATA[ + static struct snd_rawmidi_ops snd_mymidi_output_ops = { + .open = snd_mymidi_output_open, + .close = snd_mymidi_output_close, + .trigger = snd_mymidi_output_trigger, + }; +]]> + </programlisting> + </informalexample> + These callbacks are explained in the <link + linkend="rawmidi-interface-callbacks"><citetitle>Callbacks</citetitle></link> + section. + </para> + + <para> + If there are more than one substream, you should give a + unique name to each of them: + <informalexample> + <programlisting> +<![CDATA[ + struct snd_rawmidi_substream *substream; + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ +]]> + </programlisting> + </informalexample> + </para> + </section> + + <section id="rawmidi-interface-callbacks"> + <title>Callbacks</title> + + <para> + In all the callbacks, the private data that you've set for the + rawmidi device can be accessed as + substream->rmidi->private_data. + <!-- <code> isn't available before DocBook 4.3 --> + </para> + + <para> + If there is more than one port, your callbacks can determine the + port index from the struct snd_rawmidi_substream data passed to each + callback: + <informalexample> + <programlisting> +<![CDATA[ + struct snd_rawmidi_substream *substream; + int index = substream->number; +]]> + </programlisting> + </informalexample> + </para> + + <section id="rawmidi-interface-op-open"> + <title><function>open</function> callback</title> + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_open(struct snd_rawmidi_substream *substream); +]]> + </programlisting> + </informalexample> + + <para> + This is called when a substream is opened. + You can initialize the hardware here, but you shouldn't + start transmitting/receiving data yet. + </para> + </section> + + <section id="rawmidi-interface-op-close"> + <title><function>close</function> callback</title> + + <informalexample> + <programlisting> +<![CDATA[ + static int snd_xxx_close(struct snd_rawmidi_substream *substream); +]]> + </programlisting> + </informalexample> + + <para> + Guess what. + </para> + + <para> + The <function>open</function> and <function>close</function> + callbacks of a rawmidi device are serialized with a mutex, + and can sleep. + </para> + </section> + + <section id="rawmidi-interface-op-trigger-out"> + <title><function>trigger</function> callback for output + substreams</title> + + <informalexample> + <programlisting> +<![CDATA[ + static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up); +]]> + </programlisting> + </informalexample> + + <para> + This is called with a nonzero <parameter>up</parameter> + parameter when there is some data in the substream buffer that + must be transmitted. + </para> + + <para> + To read data from the buffer, call + <function>snd_rawmidi_transmit_peek</function>. It will + return the number of bytes that have been read; this will be + less than the number of bytes requested when there are no more + data in the buffer. + After the data have been transmitted successfully, call + <function>snd_rawmidi_transmit_ack</function> to remove the + data from the substream buffer: + <informalexample> + <programlisting> +<![CDATA[ + unsigned char data; + while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { + if (snd_mychip_try_to_transmit(data)) + snd_rawmidi_transmit_ack(substream, 1); + else + break; /* hardware FIFO full */ + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + If you know beforehand that the hardware will accept data, you + can use the <function>snd_rawmidi_transmit</function> function + which reads some data and removes them from the buffer at once: + <informalexample> + <programlisting> +<![CDATA[ + while (snd_mychip_transmit_possible()) { + unsigned char data; + if (snd_rawmidi_transmit(substream, &data, 1) != 1) + break; /* no more data */ + snd_mychip_transmit(data); + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + If you know beforehand how many bytes you can accept, you can + use a buffer size greater than one with the + <function>snd_rawmidi_transmit*</function> functions. + </para> + + <para> + The <function>trigger</function> callback must not sleep. If + the hardware FIFO is full before the substream buffer has been + emptied, you have to continue transmitting data later, either + in an interrupt handler, or with a timer if the hardware + doesn't have a MIDI transmit interrupt. + </para> + + <para> + The <function>trigger</function> callback is called with a + zero <parameter>up</parameter> parameter when the transmission + of data should be aborted. + </para> + </section> + + <section id="rawmidi-interface-op-trigger-in"> + <title><function>trigger</function> callback for input + substreams</title> + + <informalexample> + <programlisting> +<![CDATA[ + static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up); +]]> + </programlisting> + </informalexample> + + <para> + This is called with a nonzero <parameter>up</parameter> + parameter to enable receiving data, or with a zero + <parameter>up</parameter> parameter do disable receiving data. + </para> + + <para> + The <function>trigger</function> callback must not sleep; the + actual reading of data from the device is usually done in an + interrupt handler. + </para> + + <para> + When data reception is enabled, your interrupt handler should + call <function>snd_rawmidi_receive</function> for all received + data: + <informalexample> + <programlisting> +<![CDATA[ + void snd_mychip_midi_interrupt(...) + { + while (mychip_midi_available()) { + unsigned char data; + data = mychip_midi_read(); + snd_rawmidi_receive(substream, &data, 1); + } + } +]]> + </programlisting> + </informalexample> + </para> + </section> + + <section id="rawmidi-interface-op-drain"> + <title><function>drain</function> callback</title> + + <informalexample> + <programlisting> +<![CDATA[ + static void snd_xxx_drain(struct snd_rawmidi_substream *substream); +]]> + </programlisting> + </informalexample> + + <para> + This is only used with output substreams. This function should wait + until all data read from the substream buffer have been transmitted. + This ensures that the device can be closed and the driver unloaded + without losing data. + </para> + + <para> + This callback is optional. If you do not set + <structfield>drain</structfield> in the struct snd_rawmidi_ops + structure, ALSA will simply wait for 50 milliseconds + instead. + </para> + </section> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Miscellaneous Devices --> +<!-- ****************************************************** --> + <chapter id="misc-devices"> + <title>Miscellaneous Devices</title> + + <section id="misc-devices-opl3"> + <title>FM OPL3</title> + <para> + The FM OPL3 is still used in many chips (mainly for backward + compatibility). ALSA has a nice OPL3 FM control layer, too. The + OPL3 API is defined in + <filename><sound/opl3.h></filename>. + </para> + + <para> + FM registers can be directly accessed through the direct-FM API, + defined in <filename><sound/asound_fm.h></filename>. In + ALSA native mode, FM registers are accessed through + the Hardware-Dependant Device direct-FM extension API, whereas in + OSS compatible mode, FM registers can be accessed with the OSS + direct-FM compatible API in <filename>/dev/dmfmX</filename> device. + </para> + + <para> + To create the OPL3 component, you have two functions to + call. The first one is a constructor for the <type>opl3_t</type> + instance. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_opl3 *opl3; + snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, + integrated, &opl3); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The first argument is the card pointer, the second one is the + left port address, and the third is the right port address. In + most cases, the right port is placed at the left port + 2. + </para> + + <para> + The fourth argument is the hardware type. + </para> + + <para> + When the left and right ports have been already allocated by + the card driver, pass non-zero to the fifth argument + (<parameter>integrated</parameter>). Otherwise, the opl3 module will + allocate the specified ports by itself. + </para> + + <para> + When the accessing the hardware requires special method + instead of the standard I/O access, you can create opl3 instance + separately with <function>snd_opl3_new()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_opl3 *opl3; + snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Then set <structfield>command</structfield>, + <structfield>private_data</structfield> and + <structfield>private_free</structfield> for the private + access function, the private data and the destructor. + The l_port and r_port are not necessarily set. Only the + command must be set properly. You can retrieve the data + from the opl3->private_data field. + </para> + + <para> + After creating the opl3 instance via <function>snd_opl3_new()</function>, + call <function>snd_opl3_init()</function> to initialize the chip to the + proper state. Note that <function>snd_opl3_create()</function> always + calls it internally. + </para> + + <para> + If the opl3 instance is created successfully, then create a + hwdep device for this opl3. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_hwdep *opl3hwdep; + snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The first argument is the <type>opl3_t</type> instance you + created, and the second is the index number, usually 0. + </para> + + <para> + The third argument is the index-offset for the sequencer + client assigned to the OPL3 port. When there is an MPU401-UART, + give 1 for here (UART always takes 0). + </para> + </section> + + <section id="misc-devices-hardware-dependent"> + <title>Hardware-Dependent Devices</title> + <para> + Some chips need user-space access for special + controls or for loading the micro code. In such a case, you can + create a hwdep (hardware-dependent) device. The hwdep API is + defined in <filename><sound/hwdep.h></filename>. You can + find examples in opl3 driver or + <filename>isa/sb/sb16_csp.c</filename>. + </para> + + <para> + The creation of the <type>hwdep</type> instance is done via + <function>snd_hwdep_new()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_hwdep *hw; + snd_hwdep_new(card, "My HWDEP", 0, &hw); +]]> + </programlisting> + </informalexample> + + where the third argument is the index number. + </para> + + <para> + You can then pass any pointer value to the + <parameter>private_data</parameter>. + If you assign a private data, you should define the + destructor, too. The destructor function is set in + the <structfield>private_free</structfield> field. + + <informalexample> + <programlisting> +<![CDATA[ + struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); + hw->private_data = p; + hw->private_free = mydata_free; +]]> + </programlisting> + </informalexample> + + and the implementation of the destructor would be: + + <informalexample> + <programlisting> +<![CDATA[ + static void mydata_free(struct snd_hwdep *hw) + { + struct mydata *p = hw->private_data; + kfree(p); + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + The arbitrary file operations can be defined for this + instance. The file operators are defined in + the <parameter>ops</parameter> table. For example, assume that + this chip needs an ioctl. + + <informalexample> + <programlisting> +<![CDATA[ + hw->ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; +]]> + </programlisting> + </informalexample> + + And implement the callback functions as you like. + </para> + </section> + + <section id="misc-devices-IEC958"> + <title>IEC958 (S/PDIF)</title> + <para> + Usually the controls for IEC958 devices are implemented via + the control interface. There is a macro to compose a name string for + IEC958 controls, <function>SNDRV_CTL_NAME_IEC958()</function> + defined in <filename><include/asound.h></filename>. + </para> + + <para> + There are some standard controls for IEC958 status bits. These + controls use the type <type>SNDRV_CTL_ELEM_TYPE_IEC958</type>, + and the size of element is fixed as 4 bytes array + (value.iec958.status[x]). For the <structfield>info</structfield> + callback, you don't specify + the value field for this type (the count field must be set, + though). + </para> + + <para> + <quote>IEC958 Playback Con Mask</quote> is used to return the + bit-mask for the IEC958 status bits of consumer mode. Similarly, + <quote>IEC958 Playback Pro Mask</quote> returns the bitmask for + professional mode. They are read-only controls, and are defined + as MIXER controls (iface = + <constant>SNDRV_CTL_ELEM_IFACE_MIXER</constant>). + </para> + + <para> + Meanwhile, <quote>IEC958 Playback Default</quote> control is + defined for getting and setting the current default IEC958 + bits. Note that this one is usually defined as a PCM control + (iface = <constant>SNDRV_CTL_ELEM_IFACE_PCM</constant>), + although in some places it's defined as a MIXER control. + </para> + + <para> + In addition, you can define the control switches to + enable/disable or to set the raw bit mode. The implementation + will depend on the chip, but the control should be named as + <quote>IEC958 xxx</quote>, preferably using + the <function>SNDRV_CTL_NAME_IEC958()</function> macro. + </para> + + <para> + You can find several cases, for example, + <filename>pci/emu10k1</filename>, + <filename>pci/ice1712</filename>, or + <filename>pci/cmipci.c</filename>. + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Buffer and Memory Management --> +<!-- ****************************************************** --> + <chapter id="buffer-and-memory"> + <title>Buffer and Memory Management</title> + + <section id="buffer-and-memory-buffer-types"> + <title>Buffer Types</title> + <para> + ALSA provides several different buffer allocation functions + depending on the bus and the architecture. All these have a + consistent API. The allocation of physically-contiguous pages is + done via + <function>snd_malloc_xxx_pages()</function> function, where xxx + is the bus type. + </para> + + <para> + The allocation of pages with fallback is + <function>snd_malloc_xxx_pages_fallback()</function>. This + function tries to allocate the specified pages but if the pages + are not available, it tries to reduce the page sizes until + enough space is found. + </para> + + <para> + The release the pages, call + <function>snd_free_xxx_pages()</function> function. + </para> + + <para> + Usually, ALSA drivers try to allocate and reserve + a large contiguous physical space + at the time the module is loaded for the later use. + This is called <quote>pre-allocation</quote>. + As already written, you can call the following function at + pcm instance construction time (in the case of PCI bus). + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(pci), size, max); +]]> + </programlisting> + </informalexample> + + where <parameter>size</parameter> is the byte size to be + pre-allocated and the <parameter>max</parameter> is the maximum + size to be changed via the <filename>prealloc</filename> proc file. + The allocator will try to get an area as large as possible + within the given size. + </para> + + <para> + The second argument (type) and the third argument (device pointer) + are dependent on the bus. + In the case of the ISA bus, pass <function>snd_dma_isa_data()</function> + as the third argument with <constant>SNDRV_DMA_TYPE_DEV</constant> type. + For the continuous buffer unrelated to the bus can be pre-allocated + with <constant>SNDRV_DMA_TYPE_CONTINUOUS</constant> type and the + <function>snd_dma_continuous_data(GFP_KERNEL)</function> device pointer, + where <constant>GFP_KERNEL</constant> is the kernel allocation flag to + use. + For the PCI scatter-gather buffers, use + <constant>SNDRV_DMA_TYPE_DEV_SG</constant> with + <function>snd_dma_pci_data(pci)</function> + (see the + <link linkend="buffer-and-memory-non-contiguous"><citetitle>Non-Contiguous Buffers + </citetitle></link> section). + </para> + + <para> + Once the buffer is pre-allocated, you can use the + allocator in the <structfield>hw_params</structfield> callback: + + <informalexample> + <programlisting> +<![CDATA[ + snd_pcm_lib_malloc_pages(substream, size); +]]> + </programlisting> + </informalexample> + + Note that you have to pre-allocate to use this function. + </para> + </section> + + <section id="buffer-and-memory-external-hardware"> + <title>External Hardware Buffers</title> + <para> + Some chips have their own hardware buffers and the DMA + transfer from the host memory is not available. In such a case, + you need to either 1) copy/set the audio data directly to the + external hardware buffer, or 2) make an intermediate buffer and + copy/set the data from it to the external hardware buffer in + interrupts (or in tasklets, preferably). + </para> + + <para> + The first case works fine if the external hardware buffer is large + enough. This method doesn't need any extra buffers and thus is + more effective. You need to define the + <structfield>copy</structfield> and + <structfield>silence</structfield> callbacks for + the data transfer. However, there is a drawback: it cannot + be mmapped. The examples are GUS's GF1 PCM or emu8000's + wavetable PCM. + </para> + + <para> + The second case allows for mmap on the buffer, although you have + to handle an interrupt or a tasklet to transfer the data + from the intermediate buffer to the hardware buffer. You can find an + example in the vxpocket driver. + </para> + + <para> + Another case is when the chip uses a PCI memory-map + region for the buffer instead of the host memory. In this case, + mmap is available only on certain architectures like the Intel one. + In non-mmap mode, the data cannot be transferred as in the normal + way. Thus you need to define the <structfield>copy</structfield> and + <structfield>silence</structfield> callbacks as well, + as in the cases above. The examples are found in + <filename>rme32.c</filename> and <filename>rme96.c</filename>. + </para> + + <para> + The implementation of the <structfield>copy</structfield> and + <structfield>silence</structfield> callbacks depends upon + whether the hardware supports interleaved or non-interleaved + samples. The <structfield>copy</structfield> callback is + defined like below, a bit + differently depending whether the direction is playback or + capture: + + <informalexample> + <programlisting> +<![CDATA[ + static int playback_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); + static int capture_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); +]]> + </programlisting> + </informalexample> + </para> + + <para> + In the case of interleaved samples, the second argument + (<parameter>channel</parameter>) is not used. The third argument + (<parameter>pos</parameter>) points the + current position offset in frames. + </para> + + <para> + The meaning of the fourth argument is different between + playback and capture. For playback, it holds the source data + pointer, and for capture, it's the destination data pointer. + </para> + + <para> + The last argument is the number of frames to be copied. + </para> + + <para> + What you have to do in this callback is again different + between playback and capture directions. In the + playback case, you copy the given amount of data + (<parameter>count</parameter>) at the specified pointer + (<parameter>src</parameter>) to the specified offset + (<parameter>pos</parameter>) on the hardware buffer. When + coded like memcpy-like way, the copy would be like: + + <informalexample> + <programlisting> +<![CDATA[ + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src, + frames_to_bytes(runtime, count)); +]]> + </programlisting> + </informalexample> + </para> + + <para> + For the capture direction, you copy the given amount of + data (<parameter>count</parameter>) at the specified offset + (<parameter>pos</parameter>) on the hardware buffer to the + specified pointer (<parameter>dst</parameter>). + + <informalexample> + <programlisting> +<![CDATA[ + my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos), + frames_to_bytes(runtime, count)); +]]> + </programlisting> + </informalexample> + + Note that both the position and the amount of data are given + in frames. + </para> + + <para> + In the case of non-interleaved samples, the implementation + will be a bit more complicated. + </para> + + <para> + You need to check the channel argument, and if it's -1, copy + the whole channels. Otherwise, you have to copy only the + specified channel. Please check + <filename>isa/gus/gus_pcm.c</filename> as an example. + </para> + + <para> + The <structfield>silence</structfield> callback is also + implemented in a similar way. + + <informalexample> + <programlisting> +<![CDATA[ + static int silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count); +]]> + </programlisting> + </informalexample> + </para> + + <para> + The meanings of arguments are the same as in the + <structfield>copy</structfield> + callback, although there is no <parameter>src/dst</parameter> + argument. In the case of interleaved samples, the channel + argument has no meaning, as well as on + <structfield>copy</structfield> callback. + </para> + + <para> + The role of <structfield>silence</structfield> callback is to + set the given amount + (<parameter>count</parameter>) of silence data at the + specified offset (<parameter>pos</parameter>) on the hardware + buffer. Suppose that the data format is signed (that is, the + silent-data is 0), and the implementation using a memset-like + function would be like: + + <informalexample> + <programlisting> +<![CDATA[ + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, + frames_to_bytes(runtime, count)); +]]> + </programlisting> + </informalexample> + </para> + + <para> + In the case of non-interleaved samples, again, the + implementation becomes a bit more complicated. See, for example, + <filename>isa/gus/gus_pcm.c</filename>. + </para> + </section> + + <section id="buffer-and-memory-non-contiguous"> + <title>Non-Contiguous Buffers</title> + <para> + If your hardware supports the page table as in emu10k1 or the + buffer descriptors as in via82xx, you can use the scatter-gather + (SG) DMA. ALSA provides an interface for handling SG-buffers. + The API is provided in <filename><sound/pcm.h></filename>. + </para> + + <para> + For creating the SG-buffer handler, call + <function>snd_pcm_lib_preallocate_pages()</function> or + <function>snd_pcm_lib_preallocate_pages_for_all()</function> + with <constant>SNDRV_DMA_TYPE_DEV_SG</constant> + in the PCM constructor like other PCI pre-allocator. + You need to pass <function>snd_dma_pci_data(pci)</function>, + where pci is the struct <structname>pci_dev</structname> pointer + of the chip as well. + The <type>struct snd_sg_buf</type> instance is created as + substream->dma_private. You can cast + the pointer like: + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; +]]> + </programlisting> + </informalexample> + </para> + + <para> + Then call <function>snd_pcm_lib_malloc_pages()</function> + in the <structfield>hw_params</structfield> callback + as well as in the case of normal PCI buffer. + The SG-buffer handler will allocate the non-contiguous kernel + pages of the given size and map them onto the virtually contiguous + memory. The virtual pointer is addressed in runtime->dma_area. + The physical address (runtime->dma_addr) is set to zero, + because the buffer is physically non-contigous. + The physical address table is set up in sgbuf->table. + You can get the physical address at a certain offset via + <function>snd_pcm_sgbuf_get_addr()</function>. + </para> + + <para> + When a SG-handler is used, you need to set + <function>snd_pcm_sgbuf_ops_page</function> as + the <structfield>page</structfield> callback. + (See <link linkend="pcm-interface-operators-page-callback"> + <citetitle>page callback section</citetitle></link>.) + </para> + + <para> + To release the data, call + <function>snd_pcm_lib_free_pages()</function> in the + <structfield>hw_free</structfield> callback as usual. + </para> + </section> + + <section id="buffer-and-memory-vmalloced"> + <title>Vmalloc'ed Buffers</title> + <para> + It's possible to use a buffer allocated via + <function>vmalloc</function>, for example, for an intermediate + buffer. Since the allocated pages are not contiguous, you need + to set the <structfield>page</structfield> callback to obtain + the physical address at every offset. + </para> + + <para> + The implementation of <structfield>page</structfield> callback + would be like this: + + <informalexample> + <programlisting> +<![CDATA[ + #include <linux/vmalloc.h> + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(struct snd_pcm_substream *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } +]]> + </programlisting> + </informalexample> + </para> + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Proc Interface --> +<!-- ****************************************************** --> + <chapter id="proc-interface"> + <title>Proc Interface</title> + <para> + ALSA provides an easy interface for procfs. The proc files are + very useful for debugging. I recommend you set up proc files if + you write a driver and want to get a running status or register + dumps. The API is found in + <filename><sound/info.h></filename>. + </para> + + <para> + To create a proc file, call + <function>snd_card_proc_new()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + struct snd_info_entry *entry; + int err = snd_card_proc_new(card, "my-file", &entry); +]]> + </programlisting> + </informalexample> + + where the second argument specifies the name of the proc file to be + created. The above example will create a file + <filename>my-file</filename> under the card directory, + e.g. <filename>/proc/asound/card0/my-file</filename>. + </para> + + <para> + Like other components, the proc entry created via + <function>snd_card_proc_new()</function> will be registered and + released automatically in the card registration and release + functions. + </para> + + <para> + When the creation is successful, the function stores a new + instance in the pointer given in the third argument. + It is initialized as a text proc file for read only. To use + this proc file as a read-only text file as it is, set the read + callback with a private data via + <function>snd_info_set_text_ops()</function>. + + <informalexample> + <programlisting> +<![CDATA[ + snd_info_set_text_ops(entry, chip, my_proc_read); +]]> + </programlisting> + </informalexample> + + where the second argument (<parameter>chip</parameter>) is the + private data to be used in the callbacks. The third parameter + specifies the read buffer size and the fourth + (<parameter>my_proc_read</parameter>) is the callback function, which + is defined like + + <informalexample> + <programlisting> +<![CDATA[ + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); +]]> + </programlisting> + </informalexample> + + </para> + + <para> + In the read callback, use <function>snd_iprintf()</function> for + output strings, which works just like normal + <function>printf()</function>. For example, + + <informalexample> + <programlisting> +<![CDATA[ + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) + { + struct my_chip *chip = entry->private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + The file permissions can be changed afterwards. As default, it's + set as read only for all users. If you want to add write + permission for the user (root as default), do as follows: + + <informalexample> + <programlisting> +<![CDATA[ + entry->mode = S_IFREG | S_IRUGO | S_IWUSR; +]]> + </programlisting> + </informalexample> + + and set the write buffer size and the callback + + <informalexample> + <programlisting> +<![CDATA[ + entry->c.text.write = my_proc_write; +]]> + </programlisting> + </informalexample> + </para> + + <para> + For the write callback, you can use + <function>snd_info_get_line()</function> to get a text line, and + <function>snd_info_get_str()</function> to retrieve a string from + the line. Some examples are found in + <filename>core/oss/mixer_oss.c</filename>, core/oss/and + <filename>pcm_oss.c</filename>. + </para> + + <para> + For a raw-data proc-file, set the attributes as follows: + + <informalexample> + <programlisting> +<![CDATA[ + static struct snd_info_entry_ops my_file_io_ops = { + .read = my_file_io_read, + }; + + entry->content = SNDRV_INFO_CONTENT_DATA; + entry->private_data = chip; + entry->c.ops = &my_file_io_ops; + entry->size = 4096; + entry->mode = S_IFREG | S_IRUGO; +]]> + </programlisting> + </informalexample> + </para> + + <para> + The callback is much more complicated than the text-file + version. You need to use a low-level I/O functions such as + <function>copy_from/to_user()</function> to transfer the + data. + + <informalexample> + <programlisting> +<![CDATA[ + static long my_file_io_read(struct snd_info_entry *entry, + void *file_private_data, + struct file *file, + char *buf, + unsigned long count, + unsigned long pos) + { + long size = count; + if (pos + size > local_max_size) + size = local_max_size - pos; + if (copy_to_user(buf, local_data + pos, size)) + return -EFAULT; + return size; + } +]]> + </programlisting> + </informalexample> + </para> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Power Management --> +<!-- ****************************************************** --> + <chapter id="power-management"> + <title>Power Management</title> + <para> + If the chip is supposed to work with suspend/resume + functions, you need to add power-management code to the + driver. The additional code for power-management should be + <function>ifdef</function>'ed with + <constant>CONFIG_PM</constant>. + </para> + + <para> + If the driver <emphasis>fully</emphasis> supports suspend/resume + that is, the device can be + properly resumed to its state when suspend was called, + you can set the <constant>SNDRV_PCM_INFO_RESUME</constant> flag + in the pcm info field. Usually, this is possible when the + registers of the chip can be safely saved and restored to + RAM. If this is set, the trigger callback is called with + <constant>SNDRV_PCM_TRIGGER_RESUME</constant> after the resume + callback completes. + </para> + + <para> + Even if the driver doesn't support PM fully but + partial suspend/resume is still possible, it's still worthy to + implement suspend/resume callbacks. In such a case, applications + would reset the status by calling + <function>snd_pcm_prepare()</function> and restart the stream + appropriately. Hence, you can define suspend/resume callbacks + below but don't set <constant>SNDRV_PCM_INFO_RESUME</constant> + info flag to the PCM. + </para> + + <para> + Note that the trigger with SUSPEND can always be called when + <function>snd_pcm_suspend_all</function> is called, + regardless of the <constant>SNDRV_PCM_INFO_RESUME</constant> flag. + The <constant>RESUME</constant> flag affects only the behavior + of <function>snd_pcm_resume()</function>. + (Thus, in theory, + <constant>SNDRV_PCM_TRIGGER_RESUME</constant> isn't needed + to be handled in the trigger callback when no + <constant>SNDRV_PCM_INFO_RESUME</constant> flag is set. But, + it's better to keep it for compatibility reasons.) + </para> + <para> + In the earlier version of ALSA drivers, a common + power-management layer was provided, but it has been removed. + The driver needs to define the suspend/resume hooks according to + the bus the device is connected to. In the case of PCI drivers, the + callbacks look like below: + + <informalexample> + <programlisting> +<![CDATA[ + #ifdef CONFIG_PM + static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + { + .... /* do things for suspend */ + return 0; + } + static int snd_my_resume(struct pci_dev *pci) + { + .... /* do things for suspend */ + return 0; + } + #endif +]]> + </programlisting> + </informalexample> + </para> + + <para> + The scheme of the real suspend job is as follows. + + <orderedlist> + <listitem><para>Retrieve the card and the chip data.</para></listitem> + <listitem><para>Call <function>snd_power_change_state()</function> with + <constant>SNDRV_CTL_POWER_D3hot</constant> to change the + power status.</para></listitem> + <listitem><para>Call <function>snd_pcm_suspend_all()</function> to suspend the running PCM streams.</para></listitem> + <listitem><para>If AC97 codecs are used, call + <function>snd_ac97_suspend()</function> for each codec.</para></listitem> + <listitem><para>Save the register values if necessary.</para></listitem> + <listitem><para>Stop the hardware if necessary.</para></listitem> + <listitem><para>Disable the PCI device by calling + <function>pci_disable_device()</function>. Then, call + <function>pci_save_state()</function> at last.</para></listitem> + </orderedlist> + </para> + + <para> + A typical code would be like: + + <informalexample> + <programlisting> +<![CDATA[ + static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_pcm_suspend_all(chip->pcm); + /* (4) */ + snd_ac97_suspend(chip->ac97); + /* (5) */ + snd_mychip_save_registers(chip); + /* (6) */ + snd_mychip_stop_hardware(chip); + /* (7) */ + pci_disable_device(pci); + pci_save_state(pci); + return 0; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + The scheme of the real resume job is as follows. + + <orderedlist> + <listitem><para>Retrieve the card and the chip data.</para></listitem> + <listitem><para>Set up PCI. First, call <function>pci_restore_state()</function>. + Then enable the pci device again by calling <function>pci_enable_device()</function>. + Call <function>pci_set_master()</function> if necessary, too.</para></listitem> + <listitem><para>Re-initialize the chip.</para></listitem> + <listitem><para>Restore the saved registers if necessary.</para></listitem> + <listitem><para>Resume the mixer, e.g. calling + <function>snd_ac97_resume()</function>.</para></listitem> + <listitem><para>Restart the hardware (if any).</para></listitem> + <listitem><para>Call <function>snd_power_change_state()</function> with + <constant>SNDRV_CTL_POWER_D0</constant> to notify the processes.</para></listitem> + </orderedlist> + </para> + + <para> + A typical code would be like: + + <informalexample> + <programlisting> +<![CDATA[ + static int mychip_resume(struct pci_dev *pci) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + /* (3) */ + snd_mychip_reinit_chip(chip); + /* (4) */ + snd_mychip_restore_registers(chip); + /* (5) */ + snd_ac97_resume(chip->ac97); + /* (6) */ + snd_mychip_restart_chip(chip); + /* (7) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } +]]> + </programlisting> + </informalexample> + </para> + + <para> + As shown in the above, it's better to save registers after + suspending the PCM operations via + <function>snd_pcm_suspend_all()</function> or + <function>snd_pcm_suspend()</function>. It means that the PCM + streams are already stoppped when the register snapshot is + taken. But, remember that you don't have to restart the PCM + stream in the resume callback. It'll be restarted via + trigger call with <constant>SNDRV_PCM_TRIGGER_RESUME</constant> + when necessary. + </para> + + <para> + OK, we have all callbacks now. Let's set them up. In the + initialization of the card, make sure that you can get the chip + data from the card instance, typically via + <structfield>private_data</structfield> field, in case you + created the chip data individually. + + <informalexample> + <programlisting> +<![CDATA[ + static int __devinit snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + .... + card = snd_card_new(index[dev], id[dev], THIS_MODULE, NULL); + .... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + .... + card->private_data = chip; + .... + } +]]> + </programlisting> + </informalexample> + + When you created the chip data with + <function>snd_card_new()</function>, it's anyway accessible + via <structfield>private_data</structfield> field. + + <informalexample> + <programlisting> +<![CDATA[ + static int __devinit snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + .... + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip)); + .... + chip = card->private_data; + .... + } +]]> + </programlisting> + </informalexample> + + </para> + + <para> + If you need a space to save the registers, allocate the + buffer for it here, too, since it would be fatal + if you cannot allocate a memory in the suspend phase. + The allocated buffer should be released in the corresponding + destructor. + </para> + + <para> + And next, set suspend/resume callbacks to the pci_driver. + + <informalexample> + <programlisting> +<![CDATA[ + static struct pci_driver driver = { + .name = "My Chip", + .id_table = snd_my_ids, + .probe = snd_my_probe, + .remove = __devexit_p(snd_my_remove), + #ifdef CONFIG_PM + .suspend = snd_my_suspend, + .resume = snd_my_resume, + #endif + }; +]]> + </programlisting> + </informalexample> + </para> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Module Parameters --> +<!-- ****************************************************** --> + <chapter id="module-parameters"> + <title>Module Parameters</title> + <para> + There are standard module options for ALSA. At least, each + module should have the <parameter>index</parameter>, + <parameter>id</parameter> and <parameter>enable</parameter> + options. + </para> + + <para> + If the module supports multiple cards (usually up to + 8 = <constant>SNDRV_CARDS</constant> cards), they should be + arrays. The default initial values are defined already as + constants for easier programming: + + <informalexample> + <programlisting> +<![CDATA[ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +]]> + </programlisting> + </informalexample> + </para> + + <para> + If the module supports only a single card, they could be single + variables, instead. <parameter>enable</parameter> option is not + always necessary in this case, but it would be better to have a + dummy option for compatibility. + </para> + + <para> + The module parameters must be declared with the standard + <function>module_param()()</function>, + <function>module_param_array()()</function> and + <function>MODULE_PARM_DESC()</function> macros. + </para> + + <para> + The typical coding would be like below: + + <informalexample> + <programlisting> +<![CDATA[ + #define CARD_NAME "My Chip" + + module_param_array(index, int, NULL, 0444); + MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); + module_param_array(id, charp, NULL, 0444); + MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); + module_param_array(enable, bool, NULL, 0444); + MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); +]]> + </programlisting> + </informalexample> + </para> + + <para> + Also, don't forget to define the module description, classes, + license and devices. Especially, the recent modprobe requires to + define the module license as GPL, etc., otherwise the system is + shown as <quote>tainted</quote>. + + <informalexample> + <programlisting> +<![CDATA[ + MODULE_DESCRIPTION("My Chip"); + MODULE_LICENSE("GPL"); + MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); +]]> + </programlisting> + </informalexample> + </para> + + </chapter> + + +<!-- ****************************************************** --> +<!-- How To Put Your Driver --> +<!-- ****************************************************** --> + <chapter id="how-to-put-your-driver"> + <title>How To Put Your Driver Into ALSA Tree</title> + <section> + <title>General</title> + <para> + So far, you've learned how to write the driver codes. + And you might have a question now: how to put my own + driver into the ALSA driver tree? + Here (finally :) the standard procedure is described briefly. + </para> + + <para> + Suppose that you create a new PCI driver for the card + <quote>xyz</quote>. The card module name would be + snd-xyz. The new driver is usually put into the alsa-driver + tree, <filename>alsa-driver/pci</filename> directory in + the case of PCI cards. + Then the driver is evaluated, audited and tested + by developers and users. After a certain time, the driver + will go to the alsa-kernel tree (to the corresponding directory, + such as <filename>alsa-kernel/pci</filename>) and eventually + will be integrated into the Linux 2.6 tree (the directory would be + <filename>linux/sound/pci</filename>). + </para> + + <para> + In the following sections, the driver code is supposed + to be put into alsa-driver tree. The two cases are covered: + a driver consisting of a single source file and one consisting + of several source files. + </para> + </section> + + <section> + <title>Driver with A Single Source File</title> + <para> + <orderedlist> + <listitem> + <para> + Modify alsa-driver/pci/Makefile + </para> + + <para> + Suppose you have a file xyz.c. Add the following + two lines + <informalexample> + <programlisting> +<![CDATA[ + snd-xyz-objs := xyz.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o +]]> + </programlisting> + </informalexample> + </para> + </listitem> + + <listitem> + <para> + Create the Kconfig entry + </para> + + <para> + Add the new entry of Kconfig for your xyz driver. + <informalexample> + <programlisting> +<![CDATA[ + config SND_XYZ + tristate "Foobar XYZ" + depends on SND + select SND_PCM + help + Say Y here to include support for Foobar XYZ soundcard. + + To compile this driver as a module, choose M here: the module + will be called snd-xyz. +]]> + </programlisting> + </informalexample> + + the line, select SND_PCM, specifies that the driver xyz supports + PCM. In addition to SND_PCM, the following components are + supported for select command: + SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, + SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. + Add the select command for each supported component. + </para> + + <para> + Note that some selections imply the lowlevel selections. + For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, + AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. + You don't need to give the lowlevel selections again. + </para> + + <para> + For the details of Kconfig script, refer to the kbuild + documentation. + </para> + + </listitem> + + <listitem> + <para> + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + </para> + </listitem> + </orderedlist> + </para> + </section> + + <section> + <title>Drivers with Several Source Files</title> + <para> + Suppose that the driver snd-xyz have several source files. + They are located in the new subdirectory, + pci/xyz. + + <orderedlist> + <listitem> + <para> + Add a new directory (<filename>xyz</filename>) in + <filename>alsa-driver/pci/Makefile</filename> as below + + <informalexample> + <programlisting> +<![CDATA[ + obj-$(CONFIG_SND) += xyz/ +]]> + </programlisting> + </informalexample> + </para> + </listitem> + + <listitem> + <para> + Under the directory <filename>xyz</filename>, create a Makefile + + <example> + <title>Sample Makefile for a driver xyz</title> + <programlisting> +<![CDATA[ + ifndef SND_TOPDIR + SND_TOPDIR=../.. + endif + + include $(SND_TOPDIR)/toplevel.config + include $(SND_TOPDIR)/Makefile.conf + + snd-xyz-objs := xyz.o abc.o def.o + + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + + include $(SND_TOPDIR)/Rules.make +]]> + </programlisting> + </example> + </para> + </listitem> + + <listitem> + <para> + Create the Kconfig entry + </para> + + <para> + This procedure is as same as in the last section. + </para> + </listitem> + + <listitem> + <para> + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + </para> + </listitem> + </orderedlist> + </para> + </section> + + </chapter> + +<!-- ****************************************************** --> +<!-- Useful Functions --> +<!-- ****************************************************** --> + <chapter id="useful-functions"> + <title>Useful Functions</title> + + <section id="useful-functions-snd-printk"> + <title><function>snd_printk()</function> and friends</title> + <para> + ALSA provides a verbose version of the + <function>printk()</function> function. If a kernel config + <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this + function prints the given message together with the file name + and the line of the caller. The <constant>KERN_XXX</constant> + prefix is processed as + well as the original <function>printk()</function> does, so it's + recommended to add this prefix, e.g. + + <informalexample> + <programlisting> +<![CDATA[ + snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\n"); +]]> + </programlisting> + </informalexample> + </para> + + <para> + There are also <function>printk()</function>'s for + debugging. <function>snd_printd()</function> can be used for + general debugging purposes. If + <constant>CONFIG_SND_DEBUG</constant> is set, this function is + compiled, and works just like + <function>snd_printk()</function>. If the ALSA is compiled + without the debugging flag, it's ignored. + </para> + + <para> + <function>snd_printdd()</function> is compiled in only when + <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is set. Please note + that <constant>CONFIG_SND_DEBUG_VERBOSE</constant> is not set as default + even if you configure the alsa-driver with + <option>--with-debug=full</option> option. You need to give + explicitly <option>--with-debug=detect</option> option instead. + </para> + </section> + + <section id="useful-functions-snd-bug"> + <title><function>snd_BUG()</function></title> + <para> + It shows the <computeroutput>BUG?</computeroutput> message and + stack trace as well as <function>snd_BUG_ON</function> at the point. + It's useful to show that a fatal error happens there. + </para> + <para> + When no debug flag is set, this macro is ignored. + </para> + </section> + + <section id="useful-functions-snd-bug-on"> + <title><function>snd_BUG_ON()</function></title> + <para> + <function>snd_BUG_ON()</function> macro is similar with + <function>WARN_ON()</function> macro. For example, + + <informalexample> + <programlisting> +<![CDATA[ + snd_BUG_ON(!pointer); +]]> + </programlisting> + </informalexample> + + or it can be used as the condition, + <informalexample> + <programlisting> +<![CDATA[ + if (snd_BUG_ON(non_zero_is_bug)) + return -EINVAL; +]]> + </programlisting> + </informalexample> + + </para> + + <para> + The macro takes an conditional expression to evaluate. + When <constant>CONFIG_SND_DEBUG</constant>, is set, the + expression is actually evaluated. If it's non-zero, it shows + the warning message such as + <computeroutput>BUG? (xxx)</computeroutput> + normally followed by stack trace. It returns the evaluated + value. + When no <constant>CONFIG_SND_DEBUG</constant> is set, this + macro always returns zero. + </para> + + </section> + + </chapter> + + +<!-- ****************************************************** --> +<!-- Acknowledgments --> +<!-- ****************************************************** --> + <chapter id="acknowledgments"> + <title>Acknowledgments</title> + <para> + I would like to thank Phil Kerr for his help for improvement and + corrections of this document. + </para> + <para> + Kevin Conder reformatted the original plain-text to the + DocBook format. + </para> + <para> + Giuliano Pochini corrected typos and contributed the example codes + in the hardware constraints section. + </para> + </chapter> +</book> diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt new file mode 100644 index 0000000..ccda41b --- /dev/null +++ b/Documentation/sound/alsa/Joystick.txt @@ -0,0 +1,86 @@ +Analog Joystick Support on ALSA Drivers +======================================= + Oct. 14, 2003 + Takashi Iwai <tiwai@suse.de> + +General +------- + +First of all, you need to enable GAMEPORT support on Linux kernel for +using a joystick with the ALSA driver. For the details of gameport +support, refer to Documentation/input/joystick.txt. + +The joystick support of ALSA drivers is different between ISA and PCI +cards. In the case of ISA (PnP) cards, it's usually handled by the +independent module (ns558). Meanwhile, the ALSA PCI drivers have the +built-in gameport support. Hence, when the ALSA PCI driver is built +in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the +gameport support on that card will be (silently) disabled. + +Some adapter modules probe the physical connection of the device at +the load time. It'd be safer to plug in the joystick device before +loading the module. + + +PCI Cards +--------- + +For PCI cards, the joystick is enabled when the appropriate module +option is specified. Some drivers don't need options, and the +joystick support is always enabled. In the former ALSA version, there +was a dynamic control API for the joystick activation. It was +changed, however, to the static module options because of the system +stability and the resource management. + +The following PCI drivers support the joystick natively. + + Driver Module Option Available Values + --------------------------------------------------------------------------- + als4000 joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + au88x0 N/A N/A + azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) + ens1370 joystick 0 = disable (default), 1 = enable + ens1371 joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x200, 0x208, 0x210, 0x218 + cmipci joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + cs4281 N/A N/A + cs46xx N/A N/A + es1938 N/A N/A + es1968 joystick 0 = disable (default), 1 = enable + sonicvibes N/A N/A + trident N/A N/A + via82xx(*1) joystick 0 = disable (default), 1 = enable + ymfpci joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x201, 0x202, 0x204, 0x205(*2) + --------------------------------------------------------------------------- + + *1) VIA686A/B only + *2) With YMF744/754 chips, the port address can be chosen arbitrarily + +The following drivers don't support gameport natively, but there are +additional modules. Load the corresponding module to add the gameport +support. + + Driver Additional Module + ----------------------------- + emu10k1 emu10k1-gp + fm801 fm801-gp + ----------------------------- + +Note: the "pcigame" and "cs461x" modules are for the OSS drivers only. + These ALSA drivers (cs46xx, trident and au88x0) have the + built-in gameport support. + +As mentioned above, ALSA PCI drivers have the built-in gameport +support, so you don't have to load ns558 module. Just load "joydev" +and the appropriate adapter module (e.g. "analog"). + + +ISA Cards +--------- + +ALSA ISA drivers don't have the built-in gameport support. +Instead, you need to load "ns558" module in addition to "joydev" and +the adapter module (e.g. "analog"). diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt new file mode 100644 index 0000000..ef42c44 --- /dev/null +++ b/Documentation/sound/alsa/MIXART.txt @@ -0,0 +1,100 @@ + Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards + Digigram <alsa@digigram.com> + + +GENERAL +======= + +The miXart8 is a multichannel audio processing and mixing soundcard +that has 4 stereo audio inputs and 4 stereo audio outputs. +The miXart8AES/EBU is the same with a add-on card that offers further +4 digital stereo audio inputs and outputs. +Furthermore the add-on card offers external clock synchronisation +(AES/EBU, Word Clock, Time Code and Video Synchro) + +The mainboard has a PowerPC that offers onboard mpeg encoding and +decoding, samplerate conversions and various effects. + +The driver don't work properly at all until the certain firmwares +are loaded, i.e. no PCM nor mixer devices will appear. +Use the mixartloader that can be found in the alsa-tools package. + + +VERSION 0.1.0 +============= + +One miXart8 board will be represented as 4 alsa cards, each with 1 +stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device. +With a miXart8AES/EBU there is in addition 1 stereo digital input +'pcm1c' and 1 stereo digital output 'pcm1p' per card. + +Formats +------- +U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE +Sample rates : 8000 - 48000 Hz continuously + +Playback +-------- +For instance the playback devices are configured to have max. 4 +substreams performing hardware mixing. This could be changed to a +maximum of 24 substreams if wished. +Mono files will be played on the left and right channel. Each channel +can be muted for each stream to use 8 analog/digital outputs separately. + +Capture +------- +There is one substream per capture device. For instance only stereo +formats are supported. + +Mixer +----- +<Master> and <Master Capture> : analog volume control of playback and capture PCM. +<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream. +<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream. +<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume +and mute control. + +Rem : for best audio quality try to keep a 0 attenuation on the PCM +and AES volume controls which is set by 219 in the range from 0 to 255 +(about 86% with alsamixer) + + +NOT YET IMPLEMENTED +=================== + +- external clock support (AES/EBU, Word Clock, Time Code, Video Sync) +- MPEG audio formats +- mono record +- on-board effects and samplerate conversions +- linked streams + + +FIRMWARE +======== + +[As of 2.6.11, the firmware can be loaded automatically with hotplug + when CONFIG_FW_LOADER is set. The mixartloader is necessary only + for older versions or when you build the driver into kernel.] + +For loading the firmware automatically after the module is loaded, use +the post-install command. For example, add the following entry to +/etc/modprobe.conf for miXart driver: + + install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ + /usr/bin/mixartloader +(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to + /etc/modules.conf, instead.) + +The firmware binaries are installed on /usr/share/alsa/firmware +(or /usr/local/share/alsa/firmware, depending to the prefix option of +configure). There will be a miXart.conf file, which define the dsp image +files. + +The firmware files are copyright by Digigram SA + + +COPYRIGHT +========= + +Copyright (c) 2003 Digigram SA <alsa@digigram.com> +Distributable under GPL. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt new file mode 100644 index 0000000..022aaeb --- /dev/null +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -0,0 +1,305 @@ + NOTES ON KERNEL OSS-EMULATION + ============================= + + Jan. 22, 2004 Takashi Iwai <tiwai@suse.de> + + +Modules +======= + +ALSA provides a powerful OSS emulation on the kernel. +The OSS emulation for PCM, mixer and sequencer devices is implemented +as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. +When you need to access the OSS PCM, mixer or sequencer devices, the +corresponding module has to be loaded. + +These modules are loaded automatically when the corresponding service +is called. The alias is defined sound-service-x-y, where x and y are +the card number and the minor unit number. Usually you don't have to +define these aliases by yourself. + +Only necessary step for auto-loading of OSS modules is to define the +card alias in /etc/modprobe.conf, such as + + alias sound-slot-0 snd-emu10k1 + +As the second card, define sound-slot-1 as well. +Note that you can't use the aliased name as the target name (i.e. +"alias sound-slot-0 snd-card-0" doesn't work any more like the old +modutils). + +The currently available OSS configuration is shown in +/proc/asound/oss/sndstat. This shows in the same syntax of +/dev/sndstat, which is available on the commercial OSS driver. +On ALSA, you can symlink /dev/sndstat to this proc file. + +Please note that the devices listed in this proc file appear only +after the corresponding OSS-emulation module is loaded. Don't worry +even if "NOT ENABLED IN CONFIG" is shown in it. + + +Device Mapping +============== + +ALSA supports the following OSS device files: + + PCM: + /dev/dspX + /dev/adspX + + Mixer: + /dev/mixerX + + MIDI: + /dev/midi0X + /dev/amidi0X + + Sequencer: + /dev/sequencer + /dev/sequencer2 (aka /dev/music) + +where X is the card number from 0 to 7. + +(NOTE: Some distributions have the device files like /dev/midi0 and + /dev/midi1. They are NOT for OSS but for tclmidi, which is + a totally different thing.) + +Unlike the real OSS, ALSA cannot use the device files more than the +assigned ones. For example, the first card cannot use /dev/dsp1 or +/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. + +As seen above, PCM and MIDI may have two devices. Usually, the first +PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary +device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and +/dev/amidi, respectively. + +You can change this device mapping via the module options of +snd-pcm-oss and snd-rawmidi. In the case of PCM, the following +options are available for snd-pcm-oss: + + dsp_map PCM device number assigned to /dev/dspX + (default = 0) + adsp_map PCM device number assigned to /dev/adspX + (default = 1) + +For example, to map the third PCM device (hw:0,2) to /dev/adsp0, +define like this: + + options snd-pcm-oss adsp_map=2 + +The options take arrays. For configuring the second card, specify +two entries separated by comma. For example, to map the third PCM +device on the second card to /dev/adsp1, define like below: + + options snd-pcm-oss adsp_map=0,2 + +To change the mapping of MIDI devices, the following options are +available for snd-rawmidi: + + midi_map MIDI device number assigned to /dev/midi0X + (default = 0) + amidi_map MIDI device number assigned to /dev/amidi0X + (default = 1) + +For example, to assign the third MIDI device on the first card to +/dev/midi00, define as follows: + + options snd-rawmidi midi_map=2 + + +PCM Mode +======== + +As default, ALSA emulates the OSS PCM with so-called plugin layer, +i.e. tries to convert the sample format, rate or channels +automatically when the card doesn't support it natively. +This will lead to some problems for some applications like quake or +wine, especially if they use the card only in the MMAP mode. + +In such a case, you can change the behavior of PCM per application by +writing a command to the proc file. There is a proc file for each PCM +stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number +(zero-based), Y the PCM device number (zero-based), and 'p' is for +playback and 'c' for capture, respectively. Note that this proc file +exists only after snd-pcm-oss module is loaded. + +The command sequence has the following syntax: + + app_name fragments fragment_size [options] + +app_name is the name of application with (higher priority) or without +path. +fragments specifies the number of fragments or zero if no specific +number is given. +fragment_size is the size of fragment in bytes or zero if not given. +options is the optional parameters. The following options are +available: + + disable the application tries to open a pcm device for + this channel but does not want to use it. + direct don't use plugins + block force block open mode + non-block force non-block open mode + partial-frag write also partial fragments (affects playback only) + no-silence do not fill silence ahead to avoid clicks + +The disable option is useful when one stream direction (playback or +capture) is not handled correctly by the application although the +hardware itself does support both directions. +The direct option is used, as mentioned above, to bypass the automatic +conversion and useful for MMAP-applications. +For example, to playback the first PCM device without plugins for +quake, send a command via echo like the following: + + % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss + +While quake wants only playback, you may append the second command +to notify driver that only this direction is about to be allocated: + + % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss + +The permission of proc files depend on the module options of snd. +As default it's set as root, so you'll likely need to be superuser for +sending the command above. + +The block and non-block options are used to change the behavior of +opening the device file. + +As default, ALSA behaves as original OSS drivers, i.e. does not block +the file when it's busy. The -EBUSY error is returned in this case. + +This blocking behavior can be changed globally via nonblock_open +module option of snd-pcm-oss. For using the blocking mode as default +for OSS devices, define like the following: + + options snd-pcm-oss nonblock_open=0 + +The partial-frag and no-silence commands have been added recently. +Both commands are for optimization use only. The former command +specifies to invoke the write transfer only when the whole fragment is +filled. The latter stops writing the silence data ahead +automatically. Both are disabled as default. + +You can check the currently defined configuration by reading the proc +file. The read image can be sent to the proc file again, hence you +can save the current configuration + + % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg + +and restore it like + + % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss + +Also, for clearing all the current configuration, send "erase" command +as below: + + % echo "erase" > /proc/asound/card0/pcm0p/oss + + +Mixer Elements +============== + +Since ALSA has completely different mixer interface, the emulation of +OSS mixer is relatively complicated. ALSA builds up a mixer element +from several different ALSA (mixer) controls based on the name +string. For example, the volume element SOUND_MIXER_PCM is composed +from "PCM Playback Volume" and "PCM Playback Switch" controls for the +playback direction and from "PCM Capture Volume" and "PCM Capture +Switch" for the capture directory (if exists). When the PCM volume of +OSS is changed, all the volume and switch controls above are adjusted +automatically. + +As default, ALSA uses the following control for OSS volumes: + + OSS volume ALSA control Index + ----------------------------------------------------- + SOUND_MIXER_VOLUME Master 0 + SOUND_MIXER_BASS Tone Control - Bass 0 + SOUND_MIXER_TREBLE Tone Control - Treble 0 + SOUND_MIXER_SYNTH Synth 0 + SOUND_MIXER_PCM PCM 0 + SOUND_MIXER_SPEAKER PC Speaker 0 + SOUND_MIXER_LINE Line 0 + SOUND_MIXER_MIC Mic 0 + SOUND_MIXER_CD CD 0 + SOUND_MIXER_IMIX Monitor Mix 0 + SOUND_MIXER_ALTPCM PCM 1 + SOUND_MIXER_RECLEV (not assigned) + SOUND_MIXER_IGAIN Capture 0 + SOUND_MIXER_OGAIN Playback 0 + SOUND_MIXER_LINE1 Aux 0 + SOUND_MIXER_LINE2 Aux 1 + SOUND_MIXER_LINE3 Aux 2 + SOUND_MIXER_DIGITAL1 Digital 0 + SOUND_MIXER_DIGITAL2 Digital 1 + SOUND_MIXER_DIGITAL3 Digital 2 + SOUND_MIXER_PHONEIN Phone 0 + SOUND_MIXER_PHONEOUT Phone 1 + SOUND_MIXER_VIDEO Video 0 + SOUND_MIXER_RADIO Radio 0 + SOUND_MIXER_MONITOR Monitor 0 + +The second column is the base-string of the corresponding ALSA +control. In fact, the controls with "XXX [Playback|Capture] +[Volume|Switch]" will be checked in addition. + +The current assignment of these mixer elements is listed in the proc +file, /proc/asound/cardX/oss_mixer, which will be like the following + + VOLUME "Master" 0 + BASS "" 0 + TREBLE "" 0 + SYNTH "" 0 + PCM "PCM" 0 + ... + +where the first column is the OSS volume element, the second column +the base-string of the corresponding ALSA control, and the third the +control index. When the string is empty, it means that the +corresponding OSS control is not available. + +For changing the assignment, you can write the configuration to this +proc file. For example, to map "Wave Playback" to the PCM volume, +send the command like the following: + + % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer + +The command is exactly as same as listed in the proc file. You can +change one or more elements, one volume per line. In the last +example, both "Wave Playback Volume" and "Wave Playback Switch" will +be affected when PCM volume is changed. + +Like the case of PCM proc file, the permission of proc files depend on +the module options of snd. you'll likely need to be superuser for +sending the command above. + +As well as in the case of PCM proc file, you can save and restore the +current mixer configuration by reading and writing the whole file +image. + + +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + +Unsupported Features +==================== + +MMAP on ICE1712 driver +---------------------- +ICE1712 supports only the unconventional format, interleaved +10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap +the buffer as the conventional (mono or 2-channels, 8 or 16bit) format +on OSS. + diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt new file mode 100644 index 0000000..f738b29 --- /dev/null +++ b/Documentation/sound/alsa/Procfile.txt @@ -0,0 +1,207 @@ + Proc Files of ALSA Drivers + ========================== + Takashi Iwai <tiwai@suse.de> + +General +------- + +ALSA has its own proc tree, /proc/asound. Many useful information are +found in this tree. When you encounter a problem and need debugging, +check the files listed in the following sections. + +Each card has its subtree cardX, where X is from 0 to 7. The +card-specific files are stored in the card* subdirectories. + + +Global Information +------------------ + +cards + Shows the list of currently configured ALSA drivers, + index, the id string, short and long descriptions. + +version + Shows the version string and compile date. + +modules + Lists the module of each card + +devices + Lists the ALSA native device mappings. + +meminfo + Shows the status of allocated pages via ALSA drivers. + Appears only when CONFIG_SND_DEBUG=y. + +hwdep + Lists the currently available hwdep devices in format of + <card>-<device>: <name> + +pcm + Lists the currently available PCM devices in format of + <card>-<device>: <id>: <name> : <sub-streams> + +timer + Lists the currently available timer devices + + +oss/devices + Lists the OSS device mappings. + +oss/sndstat + Provides the output compatible with /dev/sndstat. + You can symlink this to /dev/sndstat. + + +Card Specific Files +------------------- + +The card-specific files are found in /proc/asound/card* directories. +Some drivers (e.g. cmipci) have their own proc entries for the +register dump, etc (e.g. /proc/asound/card*/cmipci shows the register +dump). These files would be really helpful for debugging. + +When PCM devices are available on this card, you can see directories +like pcm0p or pcm1c. They hold the PCM information for each PCM +stream. The number after 'pcm' is the PCM device number from 0, and +the last 'p' or 'c' means playback or capture direction. The files in +this subtree is described later. + +The status of MIDI I/O is found in midi* files. It shows the device +name and the received/transmitted bytes through the MIDI device. + +When the card is equipped with AC97 codecs, there are codec97#* +subdirectories (described later). + +When the OSS mixer emulation is enabled (and the module is loaded), +oss_mixer file appears here, too. This shows the current mapping of +OSS mixer elements to the ALSA control elements. You can change the +mapping by writing to this device. Read OSS-Emulation.txt for +details. + + +PCM Proc Files +-------------- + +card*/pcm*/info + The general information of this PCM device: card #, device #, + substreams, etc. + +card*/pcm*/xrun_debug + This file appears when CONFIG_SND_DEBUG=y. + This shows the status of xrun (= buffer overrun/xrun) debug of + ALSA PCM middle layer, as an integer from 0 to 2. The value + can be changed by writing to this file, such as + + # cat 2 > /proc/asound/card0/pcm0p/xrun_debug + + When this value is greater than 0, the driver will show the + messages to kernel log when an xrun is detected. The debug + message is shown also when the invalid H/W pointer is detected + at the update of periods (usually called from the interrupt + handler). + + When this value is greater than 1, the driver will show the + stack trace additionally. This may help the debugging. + +card*/pcm*/sub*/info + The general information of this PCM sub-stream. + +card*/pcm*/sub*/status + The current status of this PCM sub-stream, elapsed time, + H/W position, etc. + +card*/pcm*/sub*/hw_params + The hardware parameters set for this sub-stream. + +card*/pcm*/sub*/sw_params + The soft parameters set for this sub-stream. + +card*/pcm*/sub*/prealloc + The buffer pre-allocation information. + + +AC97 Codec Information +---------------------- + +card*/codec97#*/ac97#?-? + Shows the general information of this AC97 codec chip, such as + name, capabilities, set up. + +card*/codec97#0/ac97#?-?+regs + Shows the AC97 register dump. Useful for debugging. + + When CONFIG_SND_DEBUG is enabled, you can write to this file for + changing an AC97 register directly. Pass two hex numbers. + For example, + + # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs + + +USB Audio Streams +----------------- + +card*/stream* + Shows the assignment and the current status of each audio stream + of the given card. This information is very useful for debugging. + + +HD-Audio Codecs +--------------- + +card*/codec#* + Shows the general codec information and the attribute of each + widget node. + + +Sequencer Information +--------------------- + +seq/drivers + Lists the currently available ALSA sequencer drivers. + +seq/clients + Shows the list of currently available sequencer clients and + ports. The connection status and the running status are shown + in this file, too. + +seq/queues + Lists the currently allocated/running sequencer queues. + +seq/timer + Lists the currently allocated/running sequencer timers. + +seq/oss + Lists the OSS-compatible sequencer stuffs. + + +Help For Debugging? +------------------- + +When the problem is related with PCM, first try to turn on xrun_debug +mode. This will give you the kernel messages when and where xrun +happened. + +If it's really a bug, report it with the following information: + + - the name of the driver/card, show in /proc/asound/cards + - the register dump, if available (e.g. card*/cmipci) + +when it's a PCM problem, + + - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM + sub-stream directory + +when it's a mixer problem, + + - AC97 proc files, codec97#*/* files + +for USB audio/midi, + + - output of lsusb -v + - stream* files in card directory + + +The ALSA bug-tracking system is found at: + + https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt new file mode 100644 index 0000000..f5639d4 --- /dev/null +++ b/Documentation/sound/alsa/SB-Live-mixer.txt @@ -0,0 +1,356 @@ + + Sound Blaster Live mixer / default DSP code + =========================================== + + +The EMU10K1 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K1 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) IEC958 (S/PDIF) raw PCM +-------------------------- + +This PCM device (it's the 4th PCM device (index 3!) and first subdevice +(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit +little endian streams without any modifications to the digital output +(coaxial or optical). The universal interface allows the creation of up +to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would +be easy to add support for multichannel devices to the current code, +but the conversion routines exist only for stereo (2-channel streams) +at the time. + +Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. + + +2) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + + +name='Wave Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Wave Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operates +separately (they are not inside the AC97 codec). + +name='Wave Center Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? right DAC PCM slot of the AC97 codec. + +name='Wave LFE Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? left DAC PCM slot of the AC97 codec. + +name='Wave Capture Volume',index=0 +name='Wave Capture Switch',index=0 + +These controls are used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Music Playback Volume',index=0 + +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Music Capture Volume',index=0 +name='Music Capture Switch',index=0 + +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operate +separately (they are not inside the AC97 codec). + +name='Surround Capture Volume',index=0 +name='Surround Capture Switch',index=0 + +These controls are used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Center Playback Volume',index=0 + +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? right DAC PCM slot of the AC97 codec. + +name='LFE Playback Volume',index=0 + +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? left DAC PCM slot of the AC97 codec. + +name='AC97 Playback Volume',index=0 + +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result samples are forwarded to the front DAC PCM +slots of the AC97 codec. +******************************************************************************** +*** Note: This control should be zero for the standard operations, otherwise *** +*** a digital loopback is activated. *** +******************************************************************************** + +name='AC97 Capture Volume',index=0 + +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to +the standard capture PCM device). +******************************************************************************** +*** Note: This control should be 100 (maximal value), otherwise no analog *** +*** inputs of the AC97 codec can be captured (recorded). *** +******************************************************************************** + +name='IEC958 TTL Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +name='IEC958 TTL Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='Zoom Video Playback Volume',index=0 + +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +name='Zoom Video Capture Volume',index=0 + +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 LiveDrive Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +name='IEC958 LiveDrive Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='IEC958 Coaxial Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +name='IEC958 Coaxial Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line LiveDrive Playback Volume',index=0 +name='Line LiveDrive Playback Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the front +DAC PCM slots of the AC97 codec. + +name='Line LiveDrive Capture Volume',index=1 +name='Line LiveDrive Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Tone Control - Switch',index=0 + +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 + +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 + +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='IEC958 Optical Raw Playback Switch',index=0 + +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + +name='Headphone Playback Volume',index=1 + +This control attenuates the samples for the headphone output. + +name='Headphone Center Playback Switch',index=1 + +If this switch is on, then the sample for the center PCM is put to the +left headphone output (useful for SB Live cards without separate center/LFE +output). + +name='Headphone LFE Playback Switch',index=1 + +If this switch is on, then the sample for the center PCM is put to the +right headphone output (useful for SB Live cards without separate center/LFE +output). + + +3) PCM stream related controls +------------------------------ + +name='EMU10K1 PCM Volume',index 0-31 + +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + + 0 - mono, default 0xffff (no attenuation) + 1 - left, default 0xffff (no attenuation) + 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 + +This control specifies the destination - FX-bus accumulators. There are +twelve values with this mapping: + + 0 - mono, A destination (FX-bus 0-15), default 0 + 1 - mono, B destination (FX-bus 0-15), default 1 + 2 - mono, C destination (FX-bus 0-15), default 2 + 3 - mono, D destination (FX-bus 0-15), default 3 + 4 - left, A destination (FX-bus 0-15), default 0 + 5 - left, B destination (FX-bus 0-15), default 1 + 6 - left, C destination (FX-bus 0-15), default 2 + 7 - left, D destination (FX-bus 0-15), default 3 + 8 - right, A destination (FX-bus 0-15), default 0 + 9 - right, B destination (FX-bus 0-15), default 1 + 10 - right, C destination (FX-bus 0-15), default 2 + 11 - right, D destination (FX-bus 0-15), default 3 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 + +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + + 0 - mono, A destination attn, default 255 (no attenuation) + 1 - mono, B destination attn, default 255 (no attenuation) + 2 - mono, C destination attn, default 0 (mute) + 3 - mono, D destination attn, default 0 (mute) + 4 - left, A destination attn, default 255 (no attenuation) + 5 - left, B destination attn, default 0 (mute) + 6 - left, C destination attn, default 0 (mute) + 7 - left, D destination attn, default 0 (mute) + 8 - right, A destination attn, default 0 (mute) + 9 - right, B destination attn, default 255 (no attenuation) + 10 - right, C destination attn, default 0 (mute) + 11 - right, D destination attn, default 0 (mute) + + + +4) MANUALS/PATENTS: +------------------- + +ftp://opensource.creative.com/pub/doc +------------------------------------- + + Files: + LM4545.pdf AC97 Codec + + m2049.pdf The EMU10K1 Digital Audio Processor + + hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + Patent numbers: + WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) + streams + + WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + + WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + + US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + + US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + + US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + + US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) + + US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + + US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) + pool of short term memory registers + + US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt new file mode 100644 index 0000000..1b0ac06 --- /dev/null +++ b/Documentation/sound/alsa/VIA82xx-mixer.txt @@ -0,0 +1,8 @@ + + VIA82xx mixer + ============= + +On many VIA82xx boards, the 'Input Source Select' mixer control does not work. +Setting it to 'Input2' on such boards will cause recording to hang, or fail +with EIO (input/output error) via OSS emulation. This control should be left +at 'Input1' for such cards. diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt new file mode 100644 index 0000000..751d450 --- /dev/null +++ b/Documentation/sound/alsa/emu10k1-jack.txt @@ -0,0 +1,74 @@ +This document is a guide to using the emu10k1 based devices with JACK for low +latency, multichannel recording functionality. All of my recent work to allow +Linux users to use the full capabilities of their hardware has been inspired +by the kX Project. Without their work I never would have discovered the true +power of this hardware. + + http://www.kxproject.com + - Lee Revell, 2005.03.30 + +Low latency, multichannel audio with JACK and the emu10k1/emu10k2 +----------------------------------------------------------------- + +Until recently, emu10k1 users on Linux did not have access to the same low +latency, multichannel features offered by the "kX ASIO" feature of their +Windows driver. As of ALSA 1.0.9 this is no more! + +For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback +channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or +even 32 (0.66ms) frames should work well. + +The configuration is slightly more involved than on Windows, as you have to +select the correct device for JACK to use. Actually, for qjackctl users it's +fairly self explanatory - select Duplex, then for capture and playback select +the multichannel devices, set the in and out channels to 16, and the sample +rate to 48000Hz. The command line looks like this: + +/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S + +This will give you 16 input ports and 16 output ports. + +The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the +Audigy). The mapping from FX bus to physical output is described in +SB-Live-mixer.txt (or Audigy-mixer.txt). + +The 16 input ports are connected to the 16 physical inputs. Contrary to +popular belief, all emu10k1 cards are multichannel cards. Which of these +input channels have physical inputs connected to them depends on the card +model. Trial and error is highly recommended; the pinout diagrams +for the card have been reverse engineered by some enterprising kX users and are +available on the internet. Meterbridge is helpful here, and the kX forums are +packed with useful information. + +Each input port will either correspond to a digital (SPDIF) input, an analog +input, or nothing. The one exception is the SBLive! 5.1. On these devices, +the second and third input ports are wired to the center/LFE output. You will +still see 16 capture channels, but only 14 are available for recording inputs. + +This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK +ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output) +channels. + +/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards: +-------------------------------------------- +JACK Epilog FXBUS2(nr) +-------------------------------------------- +capture_1 asio14 FXBUS2(0xe) +capture_2 asio15 FXBUS2(0xf) +capture_3 asio0 FXBUS2(0x0) +~capture_4 Center EXTOUT(0x11) // mapped to by Center +~capture_5 LFE EXTOUT(0x12) // mapped to by LFE +capture_6 asio3 FXBUS2(0x3) +capture_7 asio4 FXBUS2(0x4) +capture_8 asio5 FXBUS2(0x5) +capture_9 asio6 FXBUS2(0x6) +capture_10 asio7 FXBUS2(0x7) +capture_11 asio8 FXBUS2(0x8) +capture_12 asio9 FXBUS2(0x9) +capture_13 asio10 FXBUS2(0xa) +capture_14 asio11 FXBUS2(0xb) +capture_15 asio12 FXBUS2(0xc) +capture_16 asio13 FXBUS2(0xd) +*/ + +TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt new file mode 100644 index 0000000..34e87ec --- /dev/null +++ b/Documentation/sound/alsa/hda_codec.txt @@ -0,0 +1,322 @@ +Notes on Universal Interface for Intel High Definition Audio Codec +------------------------------------------------------------------ + +Takashi Iwai <tiwai@suse.de> + + +[Still a draft version] + + +General +======= + +The snd-hda-codec module supports the generic access function for the +High Definition (HD) audio codecs. It's designed to be independent +from the controller code like ac97 codec module. The real accessors +from/to the controller must be implemented in the lowlevel driver. + +The structure of this module is similar with ac97_codec module. +Each codec chip belongs to a bus class which communicates with the +controller. + + +Initialization of Bus Instance +============================== + +The card driver has to create struct hda_bus at first. The template +struct should be filled and passed to the constructor: + +struct hda_bus_template { + void *private_data; + struct pci_dev *pci; + const char *modelname; + struct hda_bus_ops ops; +}; + +The card driver can set and use the private_data field to retrieve its +own data in callback functions. The pci field is used when the patch +needs to check the PCI subsystem IDs, so on. For non-PCI system, it +doesn't have to be set, of course. +The modelname field specifies the board's specific configuration. The +string is passed to the codec parser, and it depends on the parser how +the string is used. +These fields, private_data, pci and modelname are all optional. + +The ops field contains the callback functions as the following: + +struct hda_bus_ops { + int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); + unsigned int (*get_response)(struct hda_codec *codec); + void (*private_free)(struct hda_bus *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*pm_notify)(struct hda_codec *codec); +#endif +}; + +The command callback is called when the codec module needs to send a +VERB to the controller. It's always a single command. +The get_response callback is called when the codec requires the answer +for the last command. These two callbacks are mandatory and have to +be given. +The third, private_free callback, is optional. It's called in the +destructor to release any necessary data in the lowlevel driver. + +The pm_notify callback is available only with +CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs +to power up or may power down. The controller should check the all +belonging codecs on the bus whether they are actually powered off +(check codec->power_on), and optionally the driver may power down the +controller side, too. + +The bus instance is created via snd_hda_bus_new(). You need to pass +the card instance, the template, and the pointer to store the +resultant bus instance. + +int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, + struct hda_bus **busp); + +It returns zero if successful. A negative return value means any +error during creation. + + +Creation of Codec Instance +========================== + +Each codec chip on the board is then created on the BUS instance. +To create a codec instance, call snd_hda_codec_new(). + +int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + struct hda_codec **codecp); + +The first argument is the BUS instance, the second argument is the +address of the codec, and the last one is the pointer to store the +resultant codec instance (can be NULL if not needed). + +The codec is stored in a linked list of bus instance. You can follow +the codec list like: + + struct hda_codec *codec; + list_for_each_entry(codec, &bus->codec_list, list) { + ... + } + +The codec isn't initialized at this stage properly. The +initialization sequence is called when the controls are built later. + + +Codec Access +============ + +To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). +snd_hda_param_read() is for reading parameters. +For writing a sequence of verbs, use snd_hda_sequence_write(). + +There are variants of cached read/write, snd_hda_codec_write_cache(), +snd_hda_sequence_write_cache(). These are used for recording the +register states for the power-mangement resume. When no PM is needed, +these are equivalent with non-cached version. + +To retrieve the number of sub nodes connected to the given node, use +snd_hda_get_sub_nodes(). The connection list can be obtained via +snd_hda_get_connections() call. + +When an unsolicited event happens, pass the event via +snd_hda_queue_unsol_event() so that the codec routines will process it +later. + + +(Mixer) Controls +================ + +To create mixer controls of all codecs, call +snd_hda_build_controls(). It then builds the mixers and does +initialization stuff on each codec. + + +PCM Stuff +========= + +snd_hda_build_pcms() gives the necessary information to create PCM +streams. When it's called, each codec belonging to the bus stores +codec->num_pcms and codec->pcm_info fields. The num_pcms indicates +the number of elements in pcm_info array. The card driver is supposed +to traverse the codec linked list, read the pcm information in +pcm_info array, and build pcm instances according to them. + +The pcm_info array contains the following record: + +/* PCM information for each substream */ +struct hda_pcm_stream { + unsigned int substreams; /* number of substreams, 0 = not exist */ + unsigned int channels_min; /* min. number of channels */ + unsigned int channels_max; /* max. number of channels */ + hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ + u32 rates; /* supported rates */ + u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ + unsigned int maxbps; /* supported max. bit per sample */ + struct hda_pcm_ops ops; +}; + +/* for PCM creation */ +struct hda_pcm { + char *name; + struct hda_pcm_stream stream[2]; +}; + +The name can be passed to snd_pcm_new(). The stream field contains +the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and +capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver +should pass substreams to snd_pcm_new() for the number of substreams +to create. + +The channels_min, channels_max, rates and formats should be copied to +runtime->hw record. They and maxbps fields are used also to compute +the format value for the HDA codec and controller. Call +snd_hda_calc_stream_format() to get the format value. + +The ops field contains the following callback functions: + +struct hda_pcm_ops { + int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, + unsigned int stream_tag, unsigned int format, + struct snd_pcm_substream *substream); + int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); +}; + +All are non-NULL, so you can call them safely without NULL check. + +The open callback should be called in PCM open after runtime->hw is +set up. It may override some setting and constraints additionally. +Similarly, the close callback should be called in the PCM close. + +The prepare callback should be called in PCM prepare. This will set +up the codec chip properly for the operation. The cleanup should be +called in hw_free to clean up the configuration. + +The caller should check the return value, at least for open and +prepare callbacks. When a negative value is returned, some error +occurred. + + +Proc Files +========== + +Each codec dumps the widget node information in +/proc/asound/card*/codec#* file. This information would be really +helpful for debugging. Please provide its contents together with the +bug report. + + +Power Management +================ + +It's simple: +Call snd_hda_suspend() in the PM suspend callback. +Call snd_hda_resume() in the PM resume callback. + + +Codec Preset (Patch) +==================== + +To set up and handle the codec functionality fully, each codec may +have a codec preset (patch). It's defined in struct hda_codec_preset: + + struct hda_codec_preset { + unsigned int id; + unsigned int mask; + unsigned int subs; + unsigned int subs_mask; + unsigned int rev; + const char *name; + int (*patch)(struct hda_codec *codec); + }; + +When the codec id and codec subsystem id match with the given id and +subs fields bitwise (with bitmask mask and subs_mask), the callback +patch is called. The patch callback should initialize the codec and +set the codec->patch_ops field. This is defined as below: + + struct hda_codec_ops { + int (*build_controls)(struct hda_codec *codec); + int (*build_pcms)(struct hda_codec *codec); + int (*init)(struct hda_codec *codec); + void (*free)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + #ifdef CONFIG_PM + int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*resume)(struct hda_codec *codec); + #endif + #ifdef CONFIG_SND_HDA_POWER_SAVE + int (*check_power_status)(struct hda_codec *codec, + hda_nid_t nid); + #endif + }; + +The build_controls callback is called from snd_hda_build_controls(). +Similarly, the build_pcms callback is called from +snd_hda_build_pcms(). The init callback is called after +build_controls to initialize the hardware. +The free callback is called as a destructor. + +The unsol_event callback is called when an unsolicited event is +received. + +The suspend and resume callbacks are for power management. +They can be NULL if no special sequence is required. When the resume +callback is NULL, the driver calls the init callback and resumes the +registers from the cache. If other handling is needed, you'd need to +write your own resume callback. There, the amp values can be resumed +via + void snd_hda_codec_resume_amp(struct hda_codec *codec); +and the other codec registers via + void snd_hda_codec_resume_cache(struct hda_codec *codec); + +The check_power_status callback is called when the amp value of the +given widget NID is changed. The codec code can turn on/off the power +appropriately from this information. + +Each entry can be NULL if not necessary to be called. + + +Generic Parser +============== + +When the device doesn't match with any given presets, the widgets are +parsed via th generic parser (hda_generic.c). Its support is +limited: no multi-channel support, for example. + + +Digital I/O +=========== + +Call snd_hda_create_spdif_out_ctls() from the patch to create controls +related with SPDIF out. + + +Helper Functions +================ + +snd_hda_get_codec_name() stores the codec name on the given string. + +snd_hda_check_board_config() can be used to obtain the configuration +information matching with the device. Define the model string table +and the table with struct snd_pci_quirk entries (zero-terminated), +and pass it to the function. The function checks the modelname given +as a module parameter, and PCI subsystem IDs. If the matching entry +is found, it returns the config field value. + +snd_hda_add_new_ctls() can be used to create and add control entries. +Pass the zero-terminated array of struct snd_kcontrol_new + +Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be +used for the entry of struct snd_kcontrol_new. + +The input MUX helper callbacks for such a control are provided, too: +snd_hda_input_mux_info() and snd_hda_input_mux_put(). See +patch_realtek.c for example. diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt new file mode 100644 index 0000000..7a67ff7 --- /dev/null +++ b/Documentation/sound/alsa/hdspm.txt @@ -0,0 +1,362 @@ +Software Interface ALSA-DSP MADI Driver + +(translated from German, so no good English ;-), +2004 - winfried ritsch + + + + Full functionality has been added to the driver. Since some of + the Controls and startup-options are ALSA-Standard and only the + special Controls are described and discussed below. + + + hardware functionality: + + + Audio transmission: + + number of channels -- depends on transmission mode + + The number of channels chosen is from 1..Nmax. The reason to + use for a lower number of channels is only resource allocation, + since unused DMA channels are disabled and less memory is + allocated. So also the throughput of the PCI system can be + scaled. (Only important for low performance boards). + + Single Speed -- 1..64 channels + + (Note: Choosing the 56channel mode for transmission or as + receiver, only 56 are transmitted/received over the MADI, but + all 64 channels are available for the mixer, so channel count + for the driver) + + Double Speed -- 1..32 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 28 are transmitted/received + over the MADI, but all 32 channels are available for the mixer, + so channel count for the driver + + + Quad Speed -- 1..16 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 14 are transmitted/received + over the MADI, but all 16 channels are available for the mixer, + so channel count for the driver + + Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) + + Sample Rates -- + + Single Speed -- 32000, 44100, 48000 + + Double Speed -- 64000, 88200, 96000 (untested) + + Quad Speed -- 128000, 176400, 192000 (untested) + + access-mode -- MMAP (memory mapped), Not interleaved + (PCM_NON-INTERLEAVED) + + buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples + + fragments -- 2 + + Hardware-pointer -- 2 Modi + + + The Card supports the readout of the actual Buffer-pointer, + where DMA reads/writes. Since of the bulk mode of PCI it is only + 64 Byte accurate. SO it is not really usable for the + ALSA-mid-level functions (here the buffer-ID gives a better + result), but if MMAP is used by the application. Therefore it + can be configured at load-time with the parameter + precise-pointer. + + + (Hint: Experimenting I found that the pointer is maximum 64 to + large never to small. So if you subtract 64 you always have a + safe pointer for writing, which is used on this mode inside + ALSA. In theory now you can get now a latency as low as 16 + Samples, which is a quarter of the interrupt possibilities.) + + Precise Pointer -- off + interrupt used for pointer-calculation + + Precise Pointer -- on + hardware pointer used. + + Controller: + + + Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to + use the standard mixer-controls, since this would break most of + (especially graphic) ALSA-Mixer GUIs. So Mixer control has be + provided by a 2-dimensional controller using the + hwdep-interface. + + Also all 128+256 Peak and RMS-Meter can be accessed via the + hwdep-interface. Since it could be a performance problem always + copying and converting Peak and RMS-Levels even if you just need + one, I decided to export the hardware structure, so that of + needed some driver-guru can implement a memory-mapping of mixer + or peak-meters over ioctl, or also to do only copying and no + conversion. A test-application shows the usage of the controller. + + Latency Controls --- not implemented !!! + + + Note: Within the windows-driver the latency is accessible of a + control-panel, but buffer-sizes are controlled with ALSA from + hwparams-calls and should not be changed in run-state, I did not + implement it here. + + + System Clock -- suspended !!!! + + Name -- "System Clock Mode" + + Access -- Read Write + + Values -- "Master" "Slave" + + + !!!! This is a hardware-function but is in conflict with the + Clock-source controller, which is a kind of ALSA-standard. I + makes sense to set the card to a special mode (master at some + frequency or slave), since even not using an Audio-application + a studio should have working synchronisations setup. So use + Clock-source-controller instead !!!! + + Clock Source + + Name -- "Sample Clock Source" + + Access -- Read Write + + Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", + "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", + "Internal 96.0 kHz" + + Choose between Master at a specific Frequency and so also the + Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref" + + + !!!! This is no pure hardware function but was implemented by + ALSA by some ALSA-drivers before, so I use it also. !!! + + + Preferred Sync Ref + + Name -- "Preferred Sync Reference" + + Access -- Read Write + + Values -- "Word" "MADI" + + + Within the Auto-sync-Mode the preferred Sync Source can be + chosen. If it is not available another is used if possible. + + Note: Since MADI has a much higher bit-rate than word-clock, the + card should synchronise better in MADI Mode. But since the + RME-PLL is very good, there are almost no problems with + word-clock too. I never found a difference. + + + TX 64 channel --- + + Name -- "TX 64 channels mode" + + Access -- Read Write + + Values -- 0 1 + + Using 64-channel-modus (1) or 56-channel-modus for + MADI-transmission (0). + + + Note: This control is for output only. Input-mode is detected + automatically from hardware sending MADI. + + + Clear TMS --- + + Name -- "Clear Track Marker" + + Access -- Read Write + + Values -- 0 1 + + + Don't use to lower 5 Audio-bits on AES as additional Bits. + + + Safe Mode oder Auto Input --- + + Name -- "Safe Mode" + + Access -- Read Write + + Values -- 0 1 + + (default on) + + If on (1), then if either the optical or coaxial connection + has a failure, there is a takeover to the working one, with no + sample failure. Its only useful if you use the second as a + backup connection. + + Input --- + + Name -- "Input Select" + + Access -- Read Write + + Values -- optical coaxial + + + Choosing the Input, optical or coaxial. If Safe-mode is active, + this is the preferred Input. + +-------------- Mixer ---------------------- + + Mixer + + Name -- "Mixer" + + Access -- Read Write + + Values - <channel-number 0-127> <Value 0-65535> + + + Here as a first value the channel-index is taken to get/set the + corresponding mixer channel, where 0-63 are the input to output + fader and 64-127 the playback to outputs fader. Value 0 + is channel muted 0 and 32768 an amplification of 1. + + Chn 1-64 + + fast mixer for the ALSA-mixer utils. The diagonal of the + mixer-matrix is implemented from playback to output. + + + Line Out + + Name -- "Line Out" + + Access -- Read Write + + Values -- 0 1 + + Switching on and off the analog out, which has nothing to do + with mixing or routing. the analog outs reflects channel 63,64. + + +--- information (only read access): + + Sample Rate + + Name -- "System Sample Rate" + + Access -- Read-only + + getting the sample rate. + + + External Rate measured + + Name -- "External Rate" + + Access -- Read only + + + Should be "Autosync Rate", but Name used is + ALSA-Scheme. External Sample frequency liked used on Autosync is + reported. + + + MADI Sync Status + + Name -- "MADI Sync Lock Status" + + Access -- Read + + Values -- 0,1,2 + + MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. + + + Word Clock Sync Status + + Name -- "Word Clock Lock Status" + + Access -- Read + + Values -- 0,1,2 + + Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. + + AutoSync + + Name -- "AutoSync Reference" + + Access -- Read + + Values -- "WordClock", "MADI", "None" + + Sync-Reference is either "WordClock", "MADI" or none. + + RX 64ch --- noch nicht implementiert + + MADI-Receiver is in 64 channel mode oder 56 channel mode. + + + AB_inp --- not tested + + Used input for Auto-Input. + + + actual Buffer Position --- not implemented + + !!! this is a ALSA internal function, so no control is used !!! + + + +Calling Parameter: + + index int array (min = 1, max = 8), + "Index value for RME HDSPM interface." card-index within ALSA + + note: ALSA-standard + + id string array (min = 1, max = 8), + "ID string for RME HDSPM interface." + + note: ALSA-standard + + enable int array (min = 1, max = 8), + "Enable/disable specific HDSPM sound-cards." + + note: ALSA-standard + + precise_ptr int array (min = 1, max = 8), + "Enable precise pointer, or disable." + + note: Use only when the application supports this (which is a special case). + + line_outs_monitor int array (min = 1, max = 8), + "Send playback streams to analog outs by default." + + + note: each playback channel is mixed to the same numbered output + channel (routed). This is against the ALSA-convention, where all + channels have to be muted on after loading the driver, but was + used before on other cards, so i historically use it again) + + + + enable_monitor int array (min = 1, max = 8), + "Enable Analog Out on Channel 63/64 by default." + + note: here the analog output is enabled (but not routed).
\ No newline at end of file diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt new file mode 100644 index 0000000..9657e80 --- /dev/null +++ b/Documentation/sound/alsa/powersave.txt @@ -0,0 +1,41 @@ +Notes on Power-Saving Mode +========================== + +AC97 and HD-audio drivers have the automatic power-saving mode. +This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE +and CONFIG_SND_HDA_POWER_SAVE options, respectively. + +With the automatic power-saving, the driver turns off the codec power +appropriately when no operation is required. When no applications use +the device and/or no analog loopback is set, the power disablement is +done fully or partially. It'll save a certain power consumption, thus +good for laptops (even for desktops). + +The time-out for automatic power-off can be specified via power_save +module option of snd-ac97-codec and snd-hda-intel modules. Specify +the time-out value in seconds. 0 means to disable the automatic +power-saving. The default value of timeout is given via +CONFIG_SND_AC97_POWER_SAVE_DEFAULT and +CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 +(the minimum value) isn't recommended because many applications try to +reopen the device frequently. 10 would be a good choice for normal +operations. + +The power_save option is exported as writable. This means you can +adjust the value via sysfs on the fly. For example, to turn on the +automatic power-save mode with 10 seconds, write to +/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): + + # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + + +Note that you might hear click noise/pop when changing the power +state. Also, it often takes certain time to wake up from the +power-down to the active state. These are often hardly to fix, so +don't report extra bug reports unless you have a fix patch ;-) + +For HD-audio interface, there is another module option, +power_save_controller. This enables/disables the power-save mode of +the controller side. Setting this on may reduce a bit more power +consumption, but might result in longer wake-up time and click noise. +Try to turn it off when you experience such a thing too often. diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html new file mode 100644 index 0000000..d9776cf --- /dev/null +++ b/Documentation/sound/alsa/seq_oss.html @@ -0,0 +1,409 @@ +<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> +<HTML> +<HEAD> + <TITLE>OSS Sequencer Emulation on ALSA</TITLE> +</HEAD> +<BODY> + +<CENTER> +<H1> + +<HR WIDTH="100%"></H1></CENTER> + +<CENTER> +<H1> +OSS Sequencer Emulation on ALSA</H1></CENTER> + +<HR WIDTH="100%"> +<P>Copyright (c) 1998,1999 by Takashi Iwai +<TT><A HREF="mailto:iwai@ww.uni-erlangen.de"><iwai@ww.uni-erlangen.de></A></TT> +<P>ver.0.1.8; Nov. 16, 1999 +<H2> + +<HR WIDTH="100%"></H2> + +<H2> +1. Description</H2> +This directory contains the OSS sequencer emulation driver on ALSA. Note +that this program is still in the development state. +<P>What this does - it provides the emulation of the OSS sequencer, access +via +<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices. +The most of applications using OSS can run if the appropriate ALSA +sequencer is prepared. +<P>The following features are emulated by this driver: +<UL> +<LI> +Normal sequencer and MIDI events:</LI> + +<BR>They are converted to the ALSA sequencer events, and sent to the corresponding +port. +<LI> +Timer events:</LI> + +<BR>The timer is not selectable by ioctl. The control rate is fixed to +100 regardless of HZ. That is, even on Alpha system, a tick is always +1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>. + +<LI> +Patch loading:</LI> + +<BR>It purely depends on the synth drivers whether it's supported since +the patch loading is realized by callback to the synth driver. +<LI> +I/O controls:</LI> + +<BR>Most of controls are accepted. Some controls +are dependent on the synth driver, as well as even on original OSS.</UL> +Furthermore, you can find the following advanced features: +<UL> +<LI> +Better queue mechanism:</LI> + +<BR>The events are queued before processing them. +<LI> +Multiple applications:</LI> + +<BR>You can run two or more applications simultaneously (even for OSS sequencer)! +However, each MIDI device is exclusive - that is, if a MIDI device is opened +once by some application, other applications can't use it. No such a restriction +in synth devices. +<LI> +Real-time event processing:</LI> + +<BR>The events can be processed in real time without using out of bound +ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed +events will be processed in real-time without queued. To switch off the +real-time mode, send RELTIME 0 event. +<LI> +<TT>/proc</TT> interface:</LI> + +<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT> +at any time. In the later version, configuration will be changed via <TT>/proc</TT> +interface, too.</UL> + +<H2> +2. Installation</H2> +Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>) +and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT> +will be created. If the synth module of your sound card supports for OSS +emulation (so far, only Emu8000 driver), this module will be loaded automatically. +Otherwise, you need to load this module manually. +<P>At beginning, this module probes all the MIDI ports which have been +already connected to the sequencer. Once after that, the creation and deletion +of ports are watched by announcement mechanism of ALSA sequencer. +<P>The available synth and MIDI devices can be found in proc interface. +Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example, +if you use an AWE64 card, you'll see like the following: +<PRE> OSS sequencer emulation version 0.1.8 + ALSA client number 63 + ALSA receiver port 0 + + Number of applications: 0 + + Number of synth devices: 1 + + synth 0: [EMU8000] + type 0x1 : subtype 0x20 : voices 32 + capabilties : ioctl enabled / load_patch enabled + + Number of MIDI devices: 3 + + midi 0: [Emu8000 Port-0] ALSA port 65:0 + capability write / opened none + + midi 1: [Emu8000 Port-1] ALSA port 65:1 + capability write / opened none + + midi 2: [0: MPU-401 (UART)] ALSA port 64:0 + capability read/write / opened none</PRE> +Note that the device number may be different from the information of +<TT>/proc/asound/oss-devices</TT> +or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT> +to play via OSS sequencer emulation. +<H2> +3. Using Synthesizer Devices</H2> +Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 +and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload, +too. +<P>If the lowlevel driver supports multiple access to synth devices (like +Emu8000 driver), two or more applications are allowed to run at the same +time. +<H2> +4. Using MIDI Devices</H2> +So far, only MIDI output was tested. MIDI input was not checked at all, +but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>. +Be aware that these numbers are mostly different from the list in +<TT>/proc/asound/oss-devices</TT>. +<H2> +5. Module Options</H2> +The following module options are available: +<UL> +<LI> +<TT>maxqlen</TT></LI> + +<BR>specifies the maximum read/write queue length. This queue is private +for OSS sequencer, so that it is independent from the queue length of ALSA +sequencer. Default value is 1024. +<LI> +<TT>seq_oss_debug</TT></LI> + +<BR>specifies the debug level and accepts zero (= no debug message) or +positive integer. Default value is 0.</UL> + +<H2> +6. Queue Mechanism</H2> +OSS sequencer emulation uses an ALSA priority queue. The +events from <TT>/dev/sequencer</TT> are processed and put onto the queue +specified by module option. +<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning. +The timing events are also parsed at this moment, so that the events may +be processed in real-time. Sending an event ABSTIME 0 switches the operation +mode to real-time mode, and sending an event RELTIME 0 switches it off. +In the real-time mode, all events are dispatched immediately. +<P>The queued events are dispatched to the corresponding ALSA sequencer +ports after scheduled time by ALSA sequencer dispatcher. +<P>If the write-queue is full, the application sleeps until a certain amount +(as default one half) becomes empty in blocking mode. The synchronization +to write timing was implemented, too. +<P>The input from MIDI devices or echo-back events are stored on read FIFO +queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the +process will be awaked. + +<H2> +7. Interface to Synthesizer Device</H2> + +<H3> +7.1. Registration</H3> +To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT> +function. +<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, + snd_seq_oss_callback_t *oper, void *private_data)</PRE> +The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and +<TT>nvoices</TT> +are used for making the appropriate synth_info structure for ioctl. The +return value is an index number of this device. This index must be remembered +for unregister. If registration is failed, -errno will be returned. +<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>: +<PRE>int snd_seq_oss_synth_unregister(int index),</PRE> +where the <TT>index</TT> is the index number returned by register function. +<H3> +7.2. Callbacks</H3> +OSS synthesizer devices have capability for sample downloading and ioctls +like sample reset. In OSS emulation, these special features are realized +by using callbacks. The registration argument oper is used to specify these +callbacks. The following callback functions must be defined: +<PRE>snd_seq_oss_callback_t: + int (*open)(snd_seq_oss_arg_t *p, void *closure); + int (*close)(snd_seq_oss_arg_t *p); + int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); + int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); + int (*reset)(snd_seq_oss_arg_t *p); +Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed +to be NULL. +<P>Each callback function takes the argument type snd_seq_oss_arg_t as the +first argument. +<PRE>struct snd_seq_oss_arg_t { + int app_index; + int file_mode; + int seq_mode; + snd_seq_addr_t addr; + void *private_data; + int event_passing; +};</PRE> +The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and +<TT>seq_mode</TT> +are initialized by OSS sequencer. The <TT>app_index</TT> is the application +index which is unique to each application opening OSS sequencer. The +<TT>file_mode</TT> +is bit-flags indicating the file operation mode. See +<TT>seq_oss.h</TT> +for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In +the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used. +<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be +filled by the synth driver at open callback. The <TT>addr</TT> contains +the address of ALSA sequencer port which is assigned to this device. If +the driver allocates memory for <TT>private_data</TT>, it must be released +in close callback by itself. +<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on +/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded +as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT> +mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT> +mode checks the note above 128 and regards it as key pressure event (mainly +for Emu8000 driver). +<H4> +7.2.1. Open Callback</H4> +The <TT>open</TT> is called at each time this device is opened by an application +using OSS sequencer. This must not be NULL. Typically, the open callback +does the following procedure: +<OL> +<LI> +Allocate private data record.</LI> + +<LI> +Create an ALSA sequencer port.</LI> + +<LI> +Set the new port address on arg->addr.</LI> + +<LI> +Set the private data record pointer on arg->private_data.</LI> +</OL> +Note that the type bit-flags in port_info of this synth port must NOT contain +<TT>TYPE_MIDI_GENERIC</TT> +bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT> +bit should NOT be included, too. This is necessary to tell it from other +normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, +return -errno. +<H4> +7.2.2 Ioctl Callback</H4> +The <TT>ioctl</TT> callback is called when the sequencer receives device-specific +ioctls. The following two ioctls should be processed by this callback: +<UL> +<LI> +<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI> + +<BR>reset all samples on memory -- return 0 +<LI> +<TT>IOCTL_SYNTH_MEMAVL</TT></LI> + +<BR>return the available memory size +<LI> +<TT>FM_4OP_ENABLE</TT></LI> + +<BR>can be ignored usually</UL> +The other ioctls are processed inside the sequencer without passing to +the lowlevel driver. +<H4> +7.2.3 Load_Patch Callback</H4> +The <TT>load_patch</TT> callback is used for sample-downloading. This callback +must read the data on user-space and transfer to each device. Return 0 +if succeeded, and -errno if failed. The format argument is the patch key +in patch_info record. The buf is user-space pointer where patch_info record +is stored. The offs can be ignored. The count is total data size of this +sample data. +<H4> +7.2.4 Close Callback</H4> +The <TT>close</TT> callback is called when this device is closed by the +applicaion. If any private data was allocated in open callback, it must +be released in the close callback. The deletion of ALSA port should be +done here, too. This callback must not be NULL. +<H4> +7.2.5 Reset Callback</H4> +The <TT>reset</TT> callback is called when sequencer device is reset or +closed by applications. The callback should turn off the sounds on the +relevant port immediately, and initialize the status of the port. If this +callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the +port. +<H3> +7.3 Events</H3> +Most of the events are processed by sequencer and translated to the adequate +ALSA sequencer events, so that each synth device can receive by input_event +callback of ALSA sequencer port. The following ALSA events should be implemented +by the driver: +<BR> +<TABLE BORDER WIDTH="75%" NOSAVE > +<TR NOSAVE> +<TD NOSAVE><B>ALSA event</B></TD> + +<TD><B>Original OSS events</B></TD> +</TR> + +<TR> +<TD>NOTEON</TD> + +<TD>SEQ_NOTEON +<BR>MIDI_NOTEON</TD> +</TR> + +<TR> +<TD>NOTE</TD> + +<TD>SEQ_NOTEOFF +<BR>MIDI_NOTEOFF</TD> +</TR> + +<TR NOSAVE> +<TD NOSAVE>KEYPRESS</TD> + +<TD>MIDI_KEY_PRESSURE</TD> +</TR> + +<TR NOSAVE> +<TD>CHANPRESS</TD> + +<TD NOSAVE>SEQ_AFTERTOUCH +<BR>MIDI_CHN_PRESSURE</TD> +</TR> + +<TR NOSAVE> +<TD NOSAVE>PGMCHANGE</TD> + +<TD NOSAVE>SEQ_PGMCHANGE +<BR>MIDI_PGM_CHANGE</TD> +</TR> + +<TR> +<TD>PITCHBEND</TD> + +<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER) +<BR>MIDI_PITCH_BEND</TD> +</TR> + +<TR> +<TD>CONTROLLER</TD> + +<TD>MIDI_CTL_CHANGE +<BR>SEQ_BALANCE (with CTL_PAN)</TD> +</TR> + +<TR> +<TD>CONTROL14</TD> + +<TD>SEQ_CONTROLLER</TD> +</TR> + +<TR> +<TD>REGPARAM</TD> + +<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD> +</TR> + +<TR> +<TD>SYSEX</TD> + +<TD>SEQ_SYSEX</TD> +</TR> +</TABLE> + +<P>The most of these behavior can be realized by MIDI emulation driver +included in the Emu8000 lowlevel driver. In the future release, this module +will be independent. +<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event +type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte +packets without any modification. The lowlevel driver should process these +events appropriately. +<H2> +8. Interface to MIDI Device</H2> +Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer +ports automatically by receiving announcement from ALSA sequencer, the +MIDI devices don't need to be registered explicitly like synth devices. +However, the MIDI port_info registered to ALSA sequencer must include a group +name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or +<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>, +must be defined, too. If these conditions are not satisfied, the port is not +registered as OSS sequencer MIDI device. +<P>The events via MIDI devices are parsed in OSS sequencer and converted +to the corresponding ALSA sequencer events. The input from MIDI sequencer +is also converted to MIDI byte events by OSS sequencer. This works just +a reverse way of seq_midi module. +<H2> +9. Known Problems / TODO's</H2> + +<UL> +<LI> +Patch loading via ALSA instrument layer is not implemented yet.</LI> +</UL> + +</BODY> +</HTML> diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt new file mode 100644 index 0000000..c191955 --- /dev/null +++ b/Documentation/sound/alsa/serial-u16550.txt @@ -0,0 +1,88 @@ + + Serial UART 16450/16550 MIDI driver + =================================== + +The adaptor module parameter allows you to select either: + + 0 - Roland Soundcanvas support (default) + 1 - Midiator MS-124T support (1) + 2 - Midiator MS-124W S/A mode (2) + 3 - MS-124W M/B mode support (3) + 4 - Generic device with multiple input support (4) + +For the Midiator MS-124W, you must set the physical M-S and A-B +switches on the Midiator to match the driver mode you select. + +In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported +(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver +sends the nonstandard MIDI command sequence F5 NN, where NN is the substream +number plus 1. Roland modules use this command to switch between different +"parts", so this feature lets you treat each part as a distinct raw MIDI +substream. The driver provides no way to send F5 00 (no selection) or to not +send the F5 NN command sequence at all; perhaps it ought to. + +Usage example for simple serial converter: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200 + +Usage example for Roland SoundCanvas with 4 MIDI ports: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4 + +In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs +module parameter is automatically set to 1. The driver sends the same data to +all four MIDI Out connectors. Set the A-B switch and the speed module +parameter to match (A=19200, B=9600). + +Usage example for MS-124T, with A-B switch in A position: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \ + speed=19200 + +In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0); +the outs module parameter is automatically set to 1. The driver sends +the same data to all four MIDI Out connectors at full MIDI speed. + +Usage example for S/A mode: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2 + +In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams; +the outs module parameter is automatically set to 16. The substream +number gives a bitmask of which MIDI Out connectors the data should be +sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to +Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports. +As a special case, midiCnD0 also sends to all ports, since it is not useful +to send the data to no ports. M/B mode has extra overhead to select the MIDI +Out for each byte, so the aggregate data rate across all four MIDI Outs is +at most one byte every 520 us, as compared with the full MIDI data rate of +one byte every 320 us per port. + +Usage example for M/B mode: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3 + +The MS-124W hardware's M/A mode is currently not supported. This mode allows +the MIDI Outs to act independently at double the aggregate throughput of M/B, +but does not allow sending the same byte simultaneously to multiple MIDI Outs. +The M/A protocol requires the driver to twiddle the modem control lines under +timing constraints, so it would be a bit more complicated to implement than +the other modes. + +Midiator models other than MS-124W and MS-124T are currently not supported. +Note that the suffix letter is significant; the MS-124 and MS-124B are not +compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114. +I do have documentation (tim.mann@compaq.com) that partially covers these models, +but no units to experiment with. The MS-124W support is tested with a real unit. +The MS-124T support is untested, but should work. + +The Generic driver supports multiple input and output substreams over a single +serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the +appropriate input or output stream (depending on the data direction). +Additionally, the CTS signal is used to regulate the data flow. The number of +inputs is specified by the ins parameter. diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt new file mode 100644 index 0000000..0ebd7ea --- /dev/null +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -0,0 +1,56 @@ +ASoC currently supports the three main Digital Audio Interfaces (DAI) found on +SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. + + +AC97 +==== + + AC97 is a five wire interface commonly found on many PC sound cards. It is +now also popular in many portable devices. This DAI has a reset line and time +multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. +The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the +frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 +frame is 21uS long and is divided into 13 time slots. + +The AC97 specification can be found at :- +http://www.intel.com/design/chipsets/audio/ac97_r23.pdf + + +I2S +=== + + I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and +Rx lines are used for audio transmission, whilst the bit clock (BCLK) and +left/right clock (LRC) synchronise the link. I2S is flexible in that either the +controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock +usually varies depending on the sample rate and the master system clock +(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate +ADC and DAC LRCLKs, this allows for simultaneous capture and playback at +different sample rates. + +I2S has several different operating modes:- + + o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC + transition. + + o Left Justified - MSB is transmitted on transition of LRC. + + o Right Justified - MSB is transmitted sample size BCLKs before LRC + transition. + +PCM +=== + +PCM is another 4 wire interface, very similar to I2S, which can support a more +flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used +to synchronise the link whilst the Tx and Rx lines are used to transmit and +receive the audio data. Bit clock usually varies depending on sample rate +whilst sync runs at the sample rate. PCM also supports Time Division +Multiplexing (TDM) in that several devices can use the bus simultaneously (this +is sometimes referred to as network mode). + +Common PCM operating modes:- + + o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. + + o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt new file mode 100644 index 0000000..b130016 --- /dev/null +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -0,0 +1,51 @@ +Audio Clocking +============== + +This text describes the audio clocking terms in ASoC and digital audio in +general. Note: Audio clocking can be complex! + + +Master Clock +------------ + +Every audio subsystem is driven by a master clock (sometimes referred to as MCLK +or SYSCLK). This audio master clock can be derived from a number of sources +(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct +audio playback and capture sample rates. + +Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that +their speed can be altered by software (depending on the system use and to save +power). Other master clocks are fixed at a set frequency (i.e. crystals). + + +DAI Clocks +---------- +The Digital Audio Interface is usually driven by a Bit Clock (often referred to +as BCLK). This clock is used to drive the digital audio data across the link +between the codec and CPU. + +The DAI also has a frame clock to signal the start of each audio frame. This +clock is sometimes referred to as LRC (left right clock) or FRAME. This clock +runs at exactly the sample rate (LRC = Rate). + +Bit Clock can be generated as follows:- + +BCLK = MCLK / x + + or + +BCLK = LRC * x + + or + +BCLK = LRC * Channels * Word Size + +This relationship depends on the codec or SoC CPU in particular. In general +it is best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and word size) to save on power. + +It is also desirable to use the codec (if possible) to drive (or master) the +audio clocks as it usually gives more accurate sample rates than the CPU. + + + diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt new file mode 100644 index 0000000..1e95342 --- /dev/null +++ b/Documentation/sound/alsa/soc/codec.txt @@ -0,0 +1,198 @@ +ASoC Codec Driver +================= + +The codec driver is generic and hardware independent code that configures the +codec to provide audio capture and playback. It should contain no code that is +specific to the target platform or machine. All platform and machine specific +code should be added to the platform and machine drivers respectively. + +Each codec driver *must* provide the following features:- + + 1) Codec DAI and PCM configuration + 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 3) Mixers and audio controls + 4) Codec audio operations + +Optionally, codec drivers can also provide:- + + 5) DAPM description. + 6) DAPM event handler. + 7) DAC Digital mute control. + +Its probably best to use this guide in conjunction with the existing codec +driver code in sound/soc/codecs/ + +ASoC Codec driver breakdown +=========================== + +1 - Codec DAI and PCM configuration +----------------------------------- +Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be +registered with the core by your machine driver. + +e.g. + +struct snd_soc_codec_dai wm8731_dai = { + .name = "WM8731", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + /* pcm operations - see section 4 below */ + .ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + }, + /* DAI operations - see DAI.txt */ + .dai_ops = { + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(wm8731_dai); + + +2 - Codec control IO +-------------------- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec drivers provide +functions to read and write the codec registers along with supplying a +register cache:- + + /* IO control data and register cache */ + void *control_data; /* codec control (i2c/3wire) data */ + void *reg_cache; + +Codec read/write should do any data formatting and call the hardware +read write below to perform the IO. These functions are called by the +core and ALSA when performing DAPM or changing the mixer:- + + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + +Codec hardware IO functions - usually points to either the I2C, SPI or AC97 +read/write:- + + hw_write_t hw_write; + hw_read_t hw_read; + + +3 - Mixers and audio controls +----------------------------- +All the codec mixers and audio controls can be defined using the convenience +macros defined in soc.h. + + #define SOC_SINGLE(xname, reg, shift, mask, invert) + +Defines a single control as follows:- + + xname = Control name e.g. "Playback Volume" + reg = codec register + shift = control bit(s) offset in register + mask = control bit size(s) e.g. mask of 7 = 3 bits + invert = the control is inverted + +Other macros include:- + + #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) + +A stereo control + + #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) + +A stereo control spanning 2 registers + + #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) + +Defines an single enumerated control as follows:- + + xreg = register + xshift = control bit(s) offset in register + xmask = control bit(s) size + xtexts = pointer to array of strings that describe each setting + + #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) + +Defines a stereo enumerated control + + +4 - Codec Audio Operations +-------------------------- +The codec driver also supports the following ALSA operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); +}; + +Please refer to the ALSA driver PCM documentation for details. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm + + +5 - DAPM description. +--------------------- +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. + +Please also see the examples in other codec drivers. + + +6 - DAPM event handler +---------------------- +This function is a callback that handles codec domain PM calls and system +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. + +Power states:- + + SNDRV_CTL_POWER_D0: /* full On */ + /* vref/mid, clk and osc on, active */ + + SNDRV_CTL_POWER_D1: /* partial On */ + SNDRV_CTL_POWER_D2: /* partial On */ + + SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, inactive */ + + SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ + + +7 - Codec DAC digital mute control +---------------------------------- +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. + +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. + +i.e. + +static int wm8974_mute(struct snd_soc_codec *codec, + struct snd_soc_codec_dai *dai, int mute) +{ + u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; + if(mute) + wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); + else + wm8974_write(codec, WM8974_DAC, mute_reg); + return 0; +} diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt new file mode 100644 index 0000000..46f9684 --- /dev/null +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -0,0 +1,290 @@ +Dynamic Audio Power Management for Portable Devices +=================================================== + +1. Description +============== + +Dynamic Audio Power Management (DAPM) is designed to allow portable +Linux devices to use the minimum amount of power within the audio +subsystem at all times. It is independent of other kernel PM and as +such, can easily co-exist with the other PM systems. + +DAPM is also completely transparent to all user space applications as +all power switching is done within the ASoC core. No code changes or +recompiling are required for user space applications. DAPM makes power +switching decisions based upon any audio stream (capture/playback) +activity and audio mixer settings within the device. + +DAPM spans the whole machine. It covers power control within the entire +audio subsystem, this includes internal codec power blocks and machine +level power systems. + +There are 4 power domains within DAPM + + 1. Codec domain - VREF, VMID (core codec and audio power) + Usually controlled at codec probe/remove and suspend/resume, although + can be set at stream time if power is not needed for sidetone, etc. + + 2. Platform/Machine domain - physically connected inputs and outputs + Is platform/machine and user action specific, is configured by the + machine driver and responds to asynchronous events e.g when HP + are inserted + + 3. Path domain - audio susbsystem signal paths + Automatically set when mixer and mux settings are changed by the user. + e.g. alsamixer, amixer. + + 4. Stream domain - DACs and ADCs. + Enabled and disabled when stream playback/capture is started and + stopped respectively. e.g. aplay, arecord. + +All DAPM power switching decisions are made automatically by consulting an audio +routing map of the whole machine. This map is specific to each machine and +consists of the interconnections between every audio component (including +internal codec components). All audio components that effect power are called +widgets hereafter. + + +2. DAPM Widgets +=============== + +Audio DAPM widgets fall into a number of types:- + + o Mixer - Mixes several analog signals into a single analog signal. + o Mux - An analog switch that outputs only one of many inputs. + o PGA - A programmable gain amplifier or attenuation widget. + o ADC - Analog to Digital Converter + o DAC - Digital to Analog Converter + o Switch - An analog switch + o Input - A codec input pin + o Output - A codec output pin + o Headphone - Headphone (and optional Jack) + o Mic - Mic (and optional Jack) + o Line - Line Input/Output (and optional Jack) + o Speaker - Speaker + o Pre - Special PRE widget (exec before all others) + o Post - Special POST widget (exec after all others) + +(Widgets are defined in include/sound/soc-dapm.h) + +Widgets are usually added in the codec driver and the machine driver. There are +convenience macros defined in soc-dapm.h that can be used to quickly build a +list of widgets of the codecs and machines DAPM widgets. + +Most widgets have a name, register, shift and invert. Some widgets have extra +parameters for stream name and kcontrols. + + +2.1 Stream Domain Widgets +------------------------- + +Stream Widgets relate to the stream power domain and only consist of ADCs +(analog to digital converters) and DACs (digital to analog converters). + +Stream widgets have the following format:- + +SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), + +NOTE: the stream name must match the corresponding stream name in your codec +snd_soc_codec_dai. + +e.g. stream widgets for HiFi playback and capture + +SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), +SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), + + +2.2 Path Domain Widgets +----------------------- + +Path domain widgets have a ability to control or affect the audio signal or +audio paths within the audio subsystem. They have the following form:- + +SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) + +Any widget kcontrols can be set using the controls and num_controls members. + +e.g. Mixer widget (the kcontrols are declared first) + +/* Output Mixer */ +static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), +}; + +SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, + ARRAY_SIZE(wm8731_output_mixer_controls)), + + +2.3 Platform/Machine domain Widgets +----------------------------------- + +Machine widgets are different from codec widgets in that they don't have a +codec register bit associated with them. A machine widget is assigned to each +machine audio component (non codec) that can be independently powered. e.g. + + o Speaker Amp + o Microphone Bias + o Jack connectors + +A machine widget can have an optional call back. + +e.g. Jack connector widget for an external Mic that enables Mic Bias +when the Mic is inserted:- + +static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) +{ + gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), + + +2.4 Codec Domain +---------------- + +The codec power domain has no widgets and is handled by the codecs DAPM event +handler. This handler is called when the codec powerstate is changed wrt to any +stream event or by kernel PM events. + + +2.5 Virtual Widgets +------------------- + +Sometimes widgets exist in the codec or machine audio map that don't have any +corresponding soft power control. In this case it is necessary to create +a virtual widget - a widget with no control bits e.g. + +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), + +This can be used to merge to signal paths together in software. + +After all the widgets have been defined, they can then be added to the DAPM +subsystem individually with a call to snd_soc_dapm_new_control(). + + +3. Codec Widget Interconnections +================================ + +Widgets are connected to each other within the codec and machine by audio paths +(called interconnections). Each interconnection must be defined in order to +create a map of all audio paths between widgets. + +This is easiest with a diagram of the codec (and schematic of the machine audio +system), as it requires joining widgets together via their audio signal paths. + +e.g., from the WM8731 output mixer (wm8731.c) + +The WM8731 output mixer has 3 inputs (sources) + + 1. Line Bypass Input + 2. DAC (HiFi playback) + 3. Mic Sidetone Input + +Each input in this example has a kcontrol associated with it (defined in example +above) and is connected to the output mixer via it's kcontrol name. We can now +connect the destination widget (wrt audio signal) with it's source widgets. + + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + +So we have :- + + Destination Widget <=== Path Name <=== Source Widget + +Or:- + + Sink, Path, Source + +Or :- + + "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch". + +When there is no path name connecting widgets (e.g. a direct connection) we +pass NULL for the path name. + +Interconnections are created with a call to:- + +snd_soc_dapm_connect_input(codec, sink, path, source); + +Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +interconnections have been registered with the core. This causes the core to +scan the codec and machine so that the internal DAPM state matches the +physical state of the machine. + + +3.1 Machine Widget Interconnections +----------------------------------- +Machine widget interconnections are created in the same way as codec ones and +directly connect the codec pins to machine level widgets. + +e.g. connects the speaker out codec pins to the internal speaker. + + /* ext speaker connected to codec pins LOUT2, ROUT2 */ + {"Ext Spk", NULL , "ROUT2"}, + {"Ext Spk", NULL , "LOUT2"}, + +This allows the DAPM to power on and off pins that are connected (and in use) +and pins that are NC respectively. + + +4 Endpoint Widgets +=================== +An endpoint is a start or end point (widget) of an audio signal within the +machine and includes the codec. e.g. + + o Headphone Jack + o Internal Speaker + o Internal Mic + o Mic Jack + o Codec Pins + +When a codec pin is NC it can be marked as not used with a call to + +snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); + +The last argument is 0 for inactive and 1 for active. This way the pin and its +input widget will never be powered up and consume power. + +This also applies to machine widgets. e.g. if a headphone is connected to a +jack then the jack can be marked active. If the headphone is removed, then +the headphone jack can be marked inactive. + + +5 DAPM Widget Events +==================== + +Some widgets can register their interest with the DAPM core in PM events. +e.g. A Speaker with an amplifier registers a widget so the amplifier can be +powered only when the spk is in use. + +/* turn speaker amplifier on/off depending on use */ +static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) +{ + gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* corgi machine dapm widgets */ +static const struct snd_soc_dapm_widget wm8731_dapm_widgets = + SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + +Please see soc-dapm.h for all other widgets that support events. + + +5.1 Event types +--------------- + +The following event types are supported by event widgets. + +/* dapm event types */ +#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ +#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ +#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ +#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ +#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ +#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt new file mode 100644 index 0000000..f370e7d --- /dev/null +++ b/Documentation/sound/alsa/soc/machine.txt @@ -0,0 +1,113 @@ +ASoC Machine Driver +=================== + +The ASoC machine (or board) driver is the code that glues together the platform +and codec drivers. + +The machine driver can contain codec and platform specific code. It registers +the audio subsystem with the kernel as a platform device and is represented by +the following struct:- + +/* SoC machine */ +struct snd_soc_machine { + char *name; + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + + /* the pre and post PM functions are used to do any PM work before and + * after the codec and DAIs do any PM work. */ + int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); + int (*suspend_post)(struct platform_device *pdev, pm_message_t state); + int (*resume_pre)(struct platform_device *pdev); + int (*resume_post)(struct platform_device *pdev); + + /* machine stream operations */ + struct snd_soc_ops *ops; + + /* CPU <--> Codec DAI links */ + struct snd_soc_dai_link *dai_link; + int num_links; +}; + +probe()/remove() +---------------- +probe/remove are optional. Do any machine specific probe here. + + +suspend()/resume() +------------------ +The machine driver has pre and post versions of suspend and resume to take care +of any machine audio tasks that have to be done before or after the codec, DAIs +and DMA is suspended and resumed. Optional. + + +Machine operations +------------------ +The machine specific audio operations can be set here. Again this is optional. + + +Machine DAI Configuration +------------------------- +The machine DAI configuration glues all the codec and CPU DAIs together. It can +also be used to set up the DAI system clock and for any machine related DAI +initialisation e.g. the machine audio map can be connected to the codec audio +map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c +for examples. + +struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. + +/* corgi digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link corgi_dai = { + .name = "WM8731", + .stream_name = "WM8731", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8731_dai, + .init = corgi_wm8731_init, + .ops = &corgi_ops, +}; + +struct snd_soc_machine then sets up the machine with it's DAIs. e.g. + +/* corgi audio machine driver */ +static struct snd_soc_machine snd_soc_machine_corgi = { + .name = "Corgi", + .dai_link = &corgi_dai, + .num_links = 1, +}; + + +Machine Audio Subsystem +----------------------- + +The machine soc device glues the platform, machine and codec driver together. +Private data can also be set here. e.g. + +/* corgi audio private data */ +static struct wm8731_setup_data corgi_wm8731_setup = { + .i2c_address = 0x1b, +}; + +/* corgi audio subsystem */ +static struct snd_soc_device corgi_snd_devdata = { + .machine = &snd_soc_machine_corgi, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &corgi_wm8731_setup, +}; + + +Machine Power Map +----------------- + +The machine driver can optionally extend the codec power map and to become an +audio power map of the audio subsystem. This allows for automatic power up/down +of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack +sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for +details. + + +Machine Controls +---------------- + +Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt new file mode 100644 index 0000000..1e4c6d3 --- /dev/null +++ b/Documentation/sound/alsa/soc/overview.txt @@ -0,0 +1,86 @@ +ALSA SoC Layer +============== + +The overall project goal of the ALSA System on Chip (ASoC) layer is to +provide better ALSA support for embedded system-on-chip processors (e.g. +pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC +subsystem there was some support in the kernel for SoC audio, however it +had some limitations:- + + * Codec drivers were often tightly coupled to the underlying SoC + CPU. This is not ideal and leads to code duplication - for example, + Linux had different wm8731 drivers for 4 different SoC platforms. + + * There was no standard method to signal user initiated audio events (e.g. + Headphone/Mic insertion, Headphone/Mic detection after an insertion + event). These are quite common events on portable devices and often require + machine specific code to re-route audio, enable amps, etc., after such an + event. + + * Drivers tended to power up the entire codec when playing (or + recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There was also no support for saving + power via changing codec oversampling rates, bias currents, etc. + + +ASoC Design +=========== + +The ASoC layer is designed to address these issues and provide the following +features :- + + * Codec independence. Allows reuse of codec drivers on other platforms + and machines. + + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC + interface and codec registers it's audio interface capabilities with the + core and are subsequently matched and configured when the application + hardware parameters are known. + + * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to + its minimum power state at all times. This includes powering up/down + internal power blocks depending on the internal codec audio routing and any + active streams. + + * Pop and click reduction. Pops and clicks can be reduced by powering the + codec up/down in the correct sequence (including using digital mute). ASoC + signals the codec when to change power states. + + * Machine specific controls: Allow machines to add controls to the sound card + (e.g. volume control for speaker amplifier). + +To achieve all this, ASoC basically splits an embedded audio system into 3 +components :- + + * Codec driver: The codec driver is platform independent and contains audio + controls, audio interface capabilities, codec DAPM definition and codec IO + functions. + + * Platform driver: The platform driver contains the audio DMA engine and audio + interface drivers (e.g. I2S, AC97, PCM) for that platform. + + * Machine driver: The machine driver handles any machine specific controls and + audio events (e.g. turning on an amp at start of playback). + + +Documentation +============= + +The documentation is spilt into the following sections:- + +overview.txt: This file. + +codec.txt: Codec driver internals. + +DAI.txt: Description of Digital Audio Interface standards and how to configure +a DAI within your codec and CPU DAI drivers. + +dapm.txt: Dynamic Audio Power Management + +platform.txt: Platform audio DMA and DAI. + +machine.txt: Machine driver internals. + +pop_clicks.txt: How to minimise audio artifacts. + +clocking.txt: ASoC clocking for best power performance. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt new file mode 100644 index 0000000..b681d17 --- /dev/null +++ b/Documentation/sound/alsa/soc/platform.txt @@ -0,0 +1,58 @@ +ASoC Platform Driver +==================== + +An ASoC platform driver can be divided into audio DMA and SoC DAI configuration +and control. The platform drivers only target the SoC CPU and must have no board +specific code. + +Audio DMA +========= + +The platform DMA driver optionally supports the following ALSA operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); + int (*trigger)(struct snd_pcm_substream *, int); +}; + +The platform driver exports its DMA functionality via struct snd_soc_platform:- + +struct snd_soc_platform { + char *name; + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + + /* pcm creation and destruction */ + int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + void (*pcm_free)(struct snd_pcm *); + + /* platform stream ops */ + struct snd_pcm_ops *pcm_ops; +}; + +Please refer to the ALSA driver documentation for details of audio DMA. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm + +An example DMA driver is soc/pxa/pxa2xx-pcm.c + + +SoC DAI Drivers +=============== + +Each SoC DAI driver must provide the following features:- + + 1) Digital audio interface (DAI) description + 2) Digital audio interface configuration + 3) PCM's description + 4) SYSCLK configuration + 5) Suspend and resume (optional) + +Please see codec.txt for a description of items 1 - 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt new file mode 100644 index 0000000..e1e74da --- /dev/null +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -0,0 +1,52 @@ +Audio Pops and Clicks +===================== + +Pops and clicks are unwanted audio artifacts caused by the powering up and down +of components within the audio subsystem. This is noticeable on PCs when an +audio module is either loaded or unloaded (at module load time the sound card is +powered up and causes a popping noise on the speakers). + +Pops and clicks can be more frequent on portable systems with DAPM. This is +because the components within the subsystem are being dynamically powered +depending on the audio usage and this can subsequently cause a small pop or +click every time a component power state is changed. + + +Minimising Playback Pops and Clicks +=================================== + +Playback pops in portable audio subsystems cannot be completely eliminated +currently, however future audio codec hardware will have better pop and click +suppression. Pops can be reduced within playback by powering the audio +components in a specific order. This order is different for startup and +shutdown and follows some basic rules:- + + Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute + + Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC + +This assumes that the codec PCM output path from the DAC is via a mixer and then +a PGA (programmable gain amplifier) before being output to the speakers. + + +Minimising Capture Pops and Clicks +================================== + +Capture artifacts are somewhat easier to get rid as we can delay activating the +ADC until all the pops have occurred. This follows similar power rules to +playback in that components are powered in a sequence depending upon stream +startup or shutdown. + + Startup Order - Input PGA --> Mixers --> ADC + + Shutdown Order - ADC --> Mixers --> Input PGA + + +Zipper Noise +============ +An unwanted zipper noise can occur within the audio playback or capture stream +when a volume control is changed near its maximum gain value. The zipper noise +is heard when the gain increase or decrease changes the mean audio signal +amplitude too quickly. It can be minimised by enabling the zero cross setting +for each volume control. The ZC forces the gain change to occur when the signal +crosses the zero amplitude line. |
