From 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Sat, 16 Apr 2005 15:20:36 -0700 Subject: Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip! --- Documentation/sound/alsa/ALSA-Configuration.txt | 1505 +++++ Documentation/sound/alsa/Audigy-mixer.txt | 345 ++ Documentation/sound/alsa/Bt87x.txt | 78 + Documentation/sound/alsa/CMIPCI.txt | 242 + Documentation/sound/alsa/ControlNames.txt | 84 + .../sound/alsa/DocBook/alsa-driver-api.tmpl | 100 + .../sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6045 ++++++++++++++++++++ Documentation/sound/alsa/Joystick.txt | 86 + Documentation/sound/alsa/MIXART.txt | 100 + Documentation/sound/alsa/OSS-Emulation.txt | 297 + Documentation/sound/alsa/Procfile.txt | 191 + Documentation/sound/alsa/SB-Live-mixer.txt | 356 ++ Documentation/sound/alsa/VIA82xx-mixer.txt | 8 + Documentation/sound/alsa/hda_codec.txt | 299 + Documentation/sound/alsa/seq_oss.html | 409 ++ Documentation/sound/alsa/serial-u16550.txt | 88 + 16 files changed, 10233 insertions(+) create mode 100644 Documentation/sound/alsa/ALSA-Configuration.txt create mode 100644 Documentation/sound/alsa/Audigy-mixer.txt create mode 100644 Documentation/sound/alsa/Bt87x.txt create mode 100644 Documentation/sound/alsa/CMIPCI.txt create mode 100644 Documentation/sound/alsa/ControlNames.txt create mode 100644 Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl create mode 100644 Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl create mode 100644 Documentation/sound/alsa/Joystick.txt create mode 100644 Documentation/sound/alsa/MIXART.txt create mode 100644 Documentation/sound/alsa/OSS-Emulation.txt create mode 100644 Documentation/sound/alsa/Procfile.txt create mode 100644 Documentation/sound/alsa/SB-Live-mixer.txt create mode 100644 Documentation/sound/alsa/VIA82xx-mixer.txt create mode 100644 Documentation/sound/alsa/hda_codec.txt create mode 100644 Documentation/sound/alsa/seq_oss.html create mode 100644 Documentation/sound/alsa/serial-u16550.txt (limited to 'Documentation/sound/alsa') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt new file mode 100644 index 000000000000..71ef0498d5e0 --- /dev/null +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -0,0 +1,1505 @@ + + 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. + device_mode + - permission mask for dynamic sound device filesystem + - This is available only when DEVFS is enabled + - Default: 0666 + - E.g.: device_mode=0660 + + + 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. + + 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 7 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-ad1816a + ------------------ + + Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. + + port - port # for AD1816A chip (PnP setup) + mpu_port - port # for MPU-401 UART (PnP setup) + fm_port - port # for OPL3 (PnP setup) + irq - IRQ # for AD1816A chip (PnP setup) + mpu_irq - IRQ # for MPU-401 UART (PnP setup) + dma1 - first DMA # for AD1816A chip (PnP setup) + dma2 - second DMA # for AD1816A chip (PnP setup) + + Module supports up to 8 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) + + Module supports up to 8 cards. This module does not support autoprobe + thus main port must be specified!!! Other ports are optional. + + 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 + + Module supports autoprobe and multiple chips (max 8). + + The power-management is supported. + + Module snd-als100 + ----------------- + + Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. + + port - port # for ALS100 (SB16) chip (PnP setup) + irq - IRQ # for ALS100 (SB16) chip (PnP setup) + dma8 - 8-bit DMA # for ALS100 (SB16) chip (PnP setup) + dma16 - 16-bit DMA # for ALS100 (SB16) chip (PnP setup) + mpu_port - port # for MPU-401 UART (PnP setup) + mpu_irq - IRQ # for MPU-401 (PnP setup) + fm_port - port # for OPL3 FM (PnP setup) + + Module supports up to 8 cards, autoprobe and PnP. + + 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 + + Module supports up to 8 cards, autoprobe and PnP. + + Module snd-atiixp + ----------------- + + Module for ATI IXP 150/200/250 AC97 controllers. + + ac97_clock - AC'97 clock (defalut = 48000) + ac97_quirk - AC'97 workaround for strange hardware + See the description of intel8x0 module for details. + spdif_aclink - S/PDIF transfer over AC-link (default = 1) + + This module supports up to 8 cards and autoprobe. + + Module snd-atiixp-modem + ----------------------- + + Module for ATI IXP 150/200/250 AC97 modem controllers. + + Module supports up to 8 cards. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + 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 applicactions don't handle the SPDIF samplerate + control correctly. If you have problems regarding this, try + another ALSA compliant mixer (alsamixer works). + + Module snd-azt2320 + ------------------ + + Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). + + port - port # for AZT2320 chip (PnP setup) + wss_port - port # for WSS (PnP setup) + mpu_port - port # for MPU-401 UART (PnP setup) + fm_port - FM port # for AZT2320 chip (PnP setup) + irq - IRQ # for AZT2320 (WSS) chip (PnP setup) + mpu_irq - IRQ # for MPU-401 UART (PnP setup) + dma1 - 1st DMA # for AZT2320 (WSS) chip (PnP setup) + dma2 - 2nd DMA # for AZT2320 (WSS) chip (PnP setup) + + Module supports up to 8 cards, PnP and autoprobe. + + Module snd-azt3328 + ------------------ + + Module for sound cards based on Aztech AZF3328 PCI chip. + + joystick - Enable joystick (default off) + + Module supports up to 8 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 + + Module supports up to 8 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 + + Module supports up to 8 cards. + + + Module snd-cmi8330 + ------------------ + + Module for sound cards based on C-Media CMI8330 ISA chips. + + 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) + + Module supports up to 8 cards and autoprobe. + + Module snd-cmipci + ----------------- + + Module for C-Media CMI8338 and 8738 PCI sound cards. + + mpu_port - 0x300,0x310,0x320,0x330, 0 = disable (default) + fm_port - 0x388 (default), 0 = disable (default) + soft_ac3 - Sofware-conversion of raw SPDIF packets (model 033 only) + (default = 1) + joystick_port - Joystick port address (0 = disable, 1 = auto-detect) + + Module supports autoprobe and multiple chips (max 8). + + 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 + + Module supports up to 8 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. + + 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 + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + Module supports up to 8 cards. This module does not support autoprobe + 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. + + 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 + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + Module supports up to 8 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) + + Module supports up to 8 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 amplifer. + thinkpad - Force to enable Thinkpad's CLKRUN control. + mmap_valid - Support OSS mmap mode (default = 0). + + Module supports up to 8 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-dt019x + ----------------- + + Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP + only) + + port - Port # (PnP setup) + mpu_port - Port # for MPU-401 (PnP setup) + fm_port - Port # for FM OPL-3 (PnP setup) + irq - IRQ # (PnP setup) + mpu_irq - IRQ # for MPU-401 (PnP setup) + dma8 - DMA # (PnP setup) + + Module supports up to 8 cards. This module is enabled only with + ISA PnP support. + + 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). + + 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 + + Module supports up to 8 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] + + Module snd-emu10k1x + ------------------- + + Module for Creative Emu10k1X (SB Live Dell OEM version) + + Module supports up to 8 cards. + + Module snd-ens1370 + ------------------ + + Module for Ensoniq AudioPCI ES1370 PCI sound cards. + * SoundBlaster PCI 64 + * SoundBlaster PCI 128 + + joystick - Enable joystick (default off) + + Module supports up to 8 cards and autoprobe. + + 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 + + Module supports up to 8 cards and autoprobe. + + Module snd-es968 + ---------------- + + Module for sound cards based on ESS ES968 chip (PnP only). + + port - port # for ES968 (SB8) chip (PnP setup) + irq - IRQ # for ES968 (SB8) chip (PnP setup) + dma1 - DMA # for ES968 (SB8) chip (PnP setup) + + Module supports up to 8 cards, PnP and autoprobe. + + 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) + + Module supports up to 8 cards and autoprobe (without MPU-401 port). + + Module snd-es18xx + ----------------- + + Module for ESS AudioDrive ES-18xx sound cards. + + 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) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + Module supports up to 8 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. + + Module supports up to 8 cards and autoprobe. + + 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) + + Module supports up to 8 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) + + Module supports up to 8 cards and autoprobe. + + 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 + + Module supports up to 8 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 + + Module supports up to 8 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 + + Module supports up to 8 cards and autoprobe. + + Module snd-hda-intel + -------------------- + + Module for Intel HD Audio (ICH6, ICH6M, ICH7) + + model - force the model name + + Module supports up to 8 cards. + + 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=" 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 + w810 3-jack + + 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 + + Module snd-hdsp + --------------- + + Module for RME Hammerfall DSP audio interface(s) + + Module supports up to 8 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. + + 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 + * Hoontech SoundTrack DSP 24 + * Hoontech SoundTrack DSP 24 Value + * Hoontech SoundTrack DSP 24 Media 7.1 + * Digigram VX442 + + 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 + omni - Omni I/O support for MidiMan M-Audio Delta44/66 + cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transciever) + in msec resolution, default value is 500 (0.5 sec) + + Module supports up to 8 cards and autoprobe. Note: The consumer part + is not used with all Envy24 based cards (for example in the MidiMan Delta + serie). + + Module snd-ice1724 + ------------------ + + Module for Envy24HT (VT/ICE1724) based PCI sound cards. + * MidiMan M Audio Revolution 7.1 + * AMP Ltd AUDIO2000 + * TerraTec Aureon Sky-5.1, Space-7.1 + + model - Use the given board model, one of the following: + revo71, amp2000, prodigy71, aureon51, aureon71, + k8x800 + + Module supports up to 8 cards and autoprobe. + + Module snd-intel8x0 + ------------------- + + Module for AC'97 motherboards from Intel and compatibles. + * Intel i810/810E, i815, i820, i830, i84x, MX440 + * SiS 7012 (SiS 735) + * NVidia NForce, NForce2 + * AMD AMD768, AMD8111 + * ALi m5455 + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + ac97_quirk - AC'97 workaround for strange hardware + The following strings are accepted: + default = don't override the default setting + disable = disable the quirk + hp_only = use headphone control as master + 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. + buggy_irq - Enable workaround for buggy interrupts on some + motherboards (default off) + + Module supports autoprobe and multiple bus-master chips (max 8). + + 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 ac97_quirk option is used to enable/override the workaround + for specific devices. Some hardware have swapped output pins + between Master and Headphone, or Surround. 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 power-management is supported. + + Module snd-intel8x0m + -------------------- + + Module for Intel ICH (i8x0) chipset MC97 modems. + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + + This module supports up to 8 cards and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + 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. + + 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) + 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 + + Module supports up to 8 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. + + 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) + 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 + + Module supports up to 8 cards, autoprobe and ISA PnP. + + Module snd-korg1212 + ------------------- + + Module for Korg 1212 IO PCI card + + Module supports up to 8 cards. + + 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) + + Module supports autoprobe and multiple chips (max 8). + + 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-mixart + ----------------- + + Module for Digigram miXart8 sound cards. + + 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-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) + + Module supports multiple devices (max 8) 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-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 + + Module supports autoprobe and multiple chips (max 8). + + 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 explicity via + 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 speicfy 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: 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. + + 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 + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + Module supports up to 8 cards and ISA PnP. This module 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. + + 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. + + 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. + + 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-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-rme32 + ---------------- + + Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, + Prodif96 and Prodif Gold) sound cards. + + Module supports up to 8 cards. + + Module snd-rme96 + ---------------- + + Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. + + Module supports up to 8 cards. + + Module snd-rme9652 + ------------------ + + Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. + + precise_ptr - Enable precise pointer (doesn't work reliably). + (default = 0) + + Module supports up to 8 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. + + 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. + + 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) + + Module supports up to 8 cards and autoprobe. + + Module snd-sb16 and snd-sbawe + ----------------------------- + + Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), + SoundBlaster AWE 32 (PnP), + SoundBlaster AWE 64 PnP + + 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) + 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) + + Module supports up to 8 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. + + 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 # + + Module supports up to 8 cards. + + Module snd-sscape + ----------------- + + Module for ENSONIQ SoundScape PnP cards. + + port - Port # (PnP setup) + irq - IRQ # (PnP setup) + mpu_irq - MPU-401 IRQ # (PnP setup) + dma - DMA # (PnP setup) + + Module supports up to 8 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. + + Module supports up to 8 cards. + + Module snd-sun-cs4231 (on sparc only) + ------------------------------------- + + Module for CS4231 sound chips found on Sparcs. + + Module supports up to 8 cards. + + Module snd-wavefront + -------------------- + + Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + + 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). + use_cs4232_midi - Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) + 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. + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + Module supports up to 8 cards and ISA PnP. + + 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) + + Module supports up to 8 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 + + Module supports up to 8 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 + + pcm_channels - max channels (voices) reserved for PCM + wavetable_size - max wavetable size in kB (4-?kb) + + Module supports up to 8 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) + + This module supports up to 8 cards, autoprobe and hotplugging. + + Module snd-usb-usx2y + -------------------- + + Module for Tascam USB US-122, US-224 and US-428 devices. + + This module supports up to 8 cards, 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 (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 (defalut), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA + [VIA8233/C,8235 only] + ac97_quirk - AC'97 workaround for strange hardware + See the description of intel8x0 module for details. + + Module supports autoprobe and multiple bus-master chips (max 8). + + 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 (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. + 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=1 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 it doesn't work, try dxs_support=4. 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. + + Note: for the MPU401 on VIA823x, use snd-mpu401 driver + additonally. The mpu_port option is for VIA686 chips only. + + Module snd-via82xx-modem + ------------------------ + + Module for VIA82xx AC97 modem + + ac97_clock - AC'97 codec clock base (default 48000Hz) + + Module supports up to 8 cards. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + 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 # (1-8, default=4) + + Module supports up to 8 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) + + Module supports up to 8 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. + + Module snd-vxpocket + ------------------- + + Module for Digigram VX-Pocket VX2 PCMCIA card. + + ibl - Capture IBL size. (default = 0, minimum size) + + Module supports up to 8 cards. The module is compiled only when + PCMCIA is supported on 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. + + 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: the driver is build only when CONFIG_ISA is set. + + Module snd-vxp440 + ----------------- + + Module for Digigram VX-Pocket 440 PCMCIA card. + + ibl - Capture IBL size. (default = 0, minimum size) + + Module supports up to 8 cards. The module is compiled only when + PCMCIA is supported on kernel. + + To activate the driver via the card manager, you'll need to set + up /etc/pcmcia/vxp440.conf. See the sound/pcmcia/vx/vxp440.c. + + 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: the driver is build only when CONFIG_ISA is set. + + 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) + + Module supports autoprobe and multiple chips (max 8). + + The power-management is supported. + + Module snd-pdaudiocf + -------------------- + + Module for Sound Core PDAudioCF sound card. + + Note: the driver is build only when CONFIG_ISA is set. + + +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 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/moprobe.conf + +In this example, the interwave card is always loaded as the first card +(index 0) and ens1371 as the second (index 1). + + +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. + + +DEVFS support +============= + +The ALSA driver fully supports the devfs extension. +You should add lines below to your devfsd.conf file: + +LOOKUP snd MODLOAD ACTION snd +REGISTER ^sound/.* PERMISSIONS root.audio 660 +REGISTER ^snd/.* PERMISSIONS root.audio 660 + +Warning: These lines assume that you have the audio group in your system. + Otherwise replace audio word with another group name (root for + example). + + +Proc interfaces (/proc/asound) +============================== + +/proc/asound/card#/pcm#[cp]/oss +------------------------------- + String "erase" - erase all additional informations about OSS applications + String " []" + + - name of application with (higher priority) or without path + - number of fragments or zero if auto + - size of fragment in bytes or zero if auto + - 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 + + 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 + + +Links +===== + + ALSA project homepage + http://www.alsa-project.org + diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt new file mode 100644 index 000000000000..5132fd95e074 --- /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 acticle 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/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt new file mode 100644 index 000000000000..11edb2fd2a5a --- /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 +, 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 +big parts copied from btaudio.txt by Gerd Knorr diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt new file mode 100644 index 000000000000..4a7df771b806 --- /dev/null +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -0,0 +1,242 @@ + Brief Notes on C-Media 8738/8338 Driver + ======================================= + + Takashi Iwai + + +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 difference between two DACs. + +- The first DAC supports U8 and S16LE formats, while the second DAC + supports only S16LE. +- The seconde 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 programmin 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 As Rear" - As mentioned above, the line-in jack is used + for the rear (3th and 4th channels) output. +"Line-In As Bass" - The line-in jack is used for the bass (5th + and 6th channels) output. +"Mic As Center/LFE" - The mic jack is used for the bass output. + If this switch is on, you cannot use a microphone as a capture + source, of course. + + +Digital I/O +----------- + +The CM8x38 provides the excellent SPDIF capability with very chip +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. 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 +--------------- + +The MPU401-UART interface is enabled as default only for the first +(CMIPCI) card. You need to set module option "midi_port" properly +for the 2nd (CMIPCI) card. + +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.. + + +Joystick and Modem +------------------ + +The joystick and modem should be available by enabling the control +switch "Joystick" and "Modem" respectively. But I myself have never +tested them yet. + + +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 000000000000..5b18298e9495 --- /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: + (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 000000000000..1f3ae3e32d69 --- /dev/null +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -0,0 +1,100 @@ + + + + + + + + + + The ALSA Driver API + + + + 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. + + + + This document is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the + implied warranty of MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE. See the GNU General Public License + for more details. + + + + You should have received a copy of the GNU General Public + License along with this program; if not, write to the Free + Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, + MA 02111-1307 USA + + + + + + Management of Cards and Devices + Card Managment +!Esound/core/init.c + + Device Components +!Esound/core/device.c + + KMOD and Device File Entries +!Esound/core/sound.c + + Memory Management Helpers +!Esound/core/memory.c +!Esound/core/memalloc.c + + + PCM API + PCM Core +!Esound/core/pcm.c +!Esound/core/pcm_lib.c +!Esound/core/pcm_native.c + + PCM Format Helpers +!Esound/core/pcm_misc.c + + PCM Memory Managment +!Esound/core/pcm_memory.c + + + Control/Mixer API + General Control Interface +!Esound/core/control.c + + AC97 Codec API +!Esound/pci/ac97/ac97_codec.c +!Esound/pci/ac97/ac97_pcm.c + + + MIDI API + Raw MIDI API +!Esound/core/rawmidi.c + + MPU401-UART API +!Esound/drivers/mpu401/mpu401_uart.c + + + Proc Info API + Proc Info Interface +!Esound/core/info.c + + + Miscellaneous Functions + Hardware-Dependent Devices API +!Esound/core/hwdep.c + + ISA DMA Helpers +!Esound/core/isadma.c + + Other Helper Macros +!Iinclude/sound/core.h + + + + 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 000000000000..e789475304b6 --- /dev/null +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -0,0 +1,6045 @@ + + + + + + + + + + Writing an ALSA Driver + + Takashi + Iwai + +
+ tiwai@suse.de +
+ + + + March 6, 2005 + 0.3.4 + + + + This document describes how to write an ALSA (Advanced Linux + Sound Architecture) driver. + + + + + + Copyright (c) 2002-2004 Takashi Iwai tiwai@suse.de + + + + 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. + + + + This document is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the + implied warranty of MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE. See the GNU General Public License + for more details. + + + + You should have received a copy of the GNU General Public + License along with this program; if not, write to the Free + Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, + MA 02111-1307 USA + + + + + + + + + + Preface + + This document describes how to write an + + ALSA (Advanced Linux Sound Architecture) + driver. The document focuses mainly on the PCI soundcard. + 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. + + + + The target of this document is ones who already have enough + skill of C language and have the basic knowledge of linux + kernel programming. This document doesn't explain the general + topics of linux kernel codes and doesn't cover the detail of + implementation of each low-level driver. It describes only how is + the standard way to write a PCI sound driver on ALSA. + + + + If you are already familiar with the older ALSA ver.0.5.x, you + can check the drivers such as es1938.c or + maestro3.c which have also almost the same + code-base in the ALSA 0.5.x tree, so you can compare the differences. + + + + This document is still a draft version. Any feedbacks and + corrections, please!! + + + + + + + + + File Tree Structure + +
+ General + + The ALSA drivers are provided in the two ways. + + + + 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 codes 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 the ALSA driver on the outside of Linux + kernel like configure script, the 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 (eventually 2.6 kernel tree) once when they are + finished and confirmed to work fine. + + + + The file tree structure of ALSA driver is depicted below. Both + alsa-kernel and alsa-driver have almost the same file + structure, except for core directory. It's + named as acore in alsa-driver tree. + + + ALSA File Tree Structure + + 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 + + + +
+ +
+ core directory + + This directory contains the middle layer, that 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. + + +
+ core/oss + + + 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 core/seq/oss directory (see + + below). + +
+ +
+ core/ioctl32 + + + 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. + +
+ +
+ core/seq + + This 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 + CONFIG_SND_SEQUENCER is set in the kernel + config. + +
+ +
+ core/seq/oss + + This contains the OSS sequencer emulation codes. + +
+ +
+ core/seq/instr + + This directory contains the modules for the sequencer + instrument layer. + +
+
+ +
+ include directory + + This is the place for the public header files of ALSA drivers, + which are to be exported to the 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 reason :) + +
+ +
+ drivers directory + + This directory contains the codes shared among different drivers + on the 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 are the codes for components which are independent from + bus and cpu architectures. + + +
+ drivers/mpu401 + + The MPU401 and MPU401-UART modules are stored here. + +
+ +
+ drivers/opl3 and opl4 + + The OPL3 and OPL4 FM-synth stuff is found here. + +
+
+ +
+ i2c directory + + This contains the ALSA i2c components. + + + + Although there is a standard i2c layer on Linux, ALSA has its + own i2c codes for some cards, because the soundcard needs only a + simple operation and the standard i2c API is too complicated for + such a purpose. + + +
+ i2c/l3 + + This is a sub-directory for ARM L3 i2c. + +
+
+ +
+ synth directory + + This contains the synth middle-level modules. + + + + So far, there is only Emu8000/Emu10k1 synth driver under + synth/emux sub-directory. + +
+ +
+ pci directory + + This and its sub-directories hold the top-level card modules + for PCI soundcards and the codes specific to the PCI BUS. + + + + The drivers compiled from a single file is stored directly on + pci directory, while the drivers with several source files are + stored on its own sub-directory (e.g. emu10k1, ice1712). + +
+ +
+ isa directory + + This and its sub-directories hold the top-level card modules + for ISA soundcards. + +
+ +
+ arm, ppc, and sparc directories + + These are for the top-level card modules which are + specific to each given architecture. + +
+ +
+ usb directory + + This contains the USB-audio driver. On the latest version, the + USB MIDI driver is integrated together with usb-audio driver. + +
+ +
+ pcmcia directory + + The PCMCIA, especially PCCard drivers will go here. CardBus + drivers will be on pci directory, because its API is identical + with the standard PCI cards. + +
+ +
+ oss directory + + The OSS/Lite source files are stored here on Linux 2.6 (or + later) tree. (In the ALSA driver tarball, it's empty, of course :) + +
+ + + + + + + + Basic Flow for PCI Drivers + +
+ Outline + + The minimum flow of PCI soundcard is like the following: + + + define the PCI ID table (see the section + PCI Entries + ). + create probe() callback. + create remove() callback. + create pci_driver table which contains the three pointers above. + create init() function just calling pci_module_init() to register the pci_driver table defined above. + create exit() function to call pci_unregister_driver() function. + + +
+ +
+ Full Code Example + + The code example is shown below. Some parts are kept + unimplemented at this moment but will be filled in the + succeeding sections. The numbers in comment lines of + snd_mychip_probe() function are the + markers. + + + Basic Flow for PCI Drivers Example + + + #include + #include + #include + #include + #include + + /* module parameters (see "Module Parameters") */ + 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 */ + typedef struct snd_mychip mychip_t; + struct snd_mychip { + snd_card_t *card; + // rest of implementation will be in the section + // "PCI Resource Managements" + }; + + /* chip-specific destructor + * (see "PCI Resource Managements") + */ + static int snd_mychip_free(mychip_t *chip) + { + .... // will be implemented later... + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(snd_device_t *device) + { + mychip_t *chip = device->device_data; + return snd_mychip_free(chip); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int __devinit snd_mychip_create(snd_card_t *card, + struct pci_dev *pci, + mychip_t **rchip) + { + mychip_t *chip; + int err; + static snd_device_ops_t ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + // check PCI availability here + // (see "PCI Resource Managements") + .... + + /* allocate a chip-specific data with zero filled */ + chip = kcalloc(1, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + // rest of initialization here; will be implemented + // later, see "PCI Resource Managements" + .... + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, + chip, &ops)) < 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; + snd_card_t *card; + mychip_t *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) */ + if ((err = snd_mychip_create(card, pci, &chip)) < 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) */ + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see "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); + } +]]> + + + +
+ +
+ Constructor + + The real constructor of PCI drivers is probe callback. The + probe callback and other component-constructors which are called + from probe callback should be defined with + __devinit prefix. You + cannot use __init prefix for them, + because any PCI device could be a hotplug device. + + + + In the probe callback, the following scheme is often used. + + +
+ 1) Check and increment the device index. + + + += SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } +]]> + + + + where enable[dev] is the module option. + + + + At each time probe 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 (step + 7). + +
+ +
+ 2) Create a card instance + + + + + + + + + + The detail will be explained in the section + + Management of Cards and Components. + +
+ +
+ 3) Create a main component + + In this part, the PCI resources are allocated. + + + + + + + + The detail will be explained in the section PCI Resource + Managements. + +
+ +
+ 4) Set the driver ID and name strings. + + + +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); +]]> + + + + The driver field holds the minimal ID string of the + chip. This is referred 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. + + + + The shortname field is a string shown as more verbose + name. The longname field contains the information which is + shown in /proc/asound/cards. + +
+ +
+ 5) Create other components, such as mixer, MIDI, etc. + + Here you define the basic components such as + PCM, + mixer (e.g. AC97), + MIDI (e.g. MPU-401), + and other interfaces. + Also, if you want a proc + file, define it here, too. + +
+ +
+ 6) Register the card instance. + + + + + + + + + + Will be explained in the section Management + of Cards and Components, too. + +
+ +
+ 7) Set the PCI driver data and return zero. + + + + + + + + In the above, the card record is stored. This pointer is + referred in the remove callback and power-management + callbacks, too. + +
+
+ +
+ Destructor + + The destructor, remove callback, simply releases the card + instance. Then the ALSA middle layer will release all the + attached components automatically. + + + + It would be typically like the following: + + + + + + + + The above code assumes that the card pointer is set to the PCI + driver data. + +
+ +
+ Header Files + + For the above example, at least the following include files + are necessary. + + + + + #include + #include + #include + #include + #include +]]> + + + + where the last one is necessary only when module options are + defined in the source file. If the codes are split to several + files, the file without module options don't need them. + + + + In addition to them, you'll need + <linux/interrupt.h> for the interrupt + handling, and <asm/io.h> for the i/o + access. If you use mdelay() or + udelay() functions, you'll need to include + <linux/delay.h>, too. + + + + The ALSA interfaces like PCM or control API are defined in other + header files as <sound/xxx.h>. + They have to be included after + <sound/core.h>. + + +
+ + + + + + + + Management of Cards and Components + +
+ Card Instance + + For each soundcard, a card record must be allocated. + + + + A card record is the headquarters of the soundcard. It manages + the list of whole 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 proper releases of resources at + destruction. + + + + As mentioned above, to create a card instance, call + snd_card_new(). + + + + + + + + + + The function takes four arguments, the card-index number, the + id string, the module pointer (usually + THIS_MODULE), + and the size of extra-data space. The last argument is used to + allocate card->private_data for the + chip-specific data. Note that this data + is allocated by + snd_card_new(). + +
+ +
+ Components + + After the card is created, you can attach the components + (devices) to the card instance. On ALSA driver, a component is + represented as a snd_device_t object. + A component can be a PCM instance, a control interface, a raw + MIDI interface, etc. Each of such instances has one component + entry. + + + + A component can be created via + snd_device_new() function. + + + + + + + + + + This takes the card pointer, the device-level + (SNDRV_DEV_XXX), the data pointer, and the + callback pointers (&ops). The + device-level defines the type of components and the order of + registration and de-registration. For most of components, the + device-level is already defined. For a user-defined component, + you can use SNDRV_DEV_LOWLEVEL. + + + + 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 identifier + (chip in the above example) for the + instance. + + + + Each ALSA pre-defined component such as ac97 or pcm calls + snd_device_new() 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. + + + + If you would like to create your own component, you need to + set the destructor function to dev_free callback in + ops, so that it can be released + automatically via snd_card_free(). The + example will be shown later as an implementation of a + chip-specific data. + +
+ +
+ Chip-Specific Data + + The chip-specific information, e.g. the i/o port address, its + resource pointer, or the irq number, is stored in the + chip-specific record. + Usually, the chip-specific record is typedef'ed as + xxx_t like the following: + + + + + + + + + + In general, there are two ways to allocate the chip record. + + +
+ 1. Allocating via <function>snd_card_new()</function>. + + As mentioned above, you can pass the extra-data-length to the 4th argument of snd_card_new(), i.e. + + + + + + + + whether mychip_t is the type of the chip record. + + + + In return, the allocated record can be accessed as + + + +private_data; +]]> + + + + With this method, you don't have to allocate twice. + The record is released together with the card instance. + +
+ +
+ 2. Allocating an extra device. + + + After allocating a card instance via + snd_card_new() (with + NULL on the 4th arg), call + kcalloc(). + + + + + + + + + + The chip record should have the field to hold the card + pointer at least, + + + + + + + + + + Then, set the card pointer in the returned chip instance. + + + +card = card; +]]> + + + + + + Next, initialize the fields, and register this chip + record as a low-level device with a specified + ops, + + + + + + + + snd_mychip_dev_free() is the + device-destructor function, which will call the real + destructor. + + + + + +device_data; + return snd_mychip_free(chip); + } +]]> + + + + where snd_mychip_free() is the real destructor. + +
+
+ +
+ Registration and Release + + After all components are assigned, register the card instance + by calling snd_card_register(). The access + to the device files are enabled at this point. That is, before + snd_card_register() is called, the + components are safely inaccessible from external side. If this + call fails, exit the probe function after releasing the card via + snd_card_free(). + + + + For releasing the card instance, you can call simply + snd_card_free(). As already mentioned, all + components are released automatically by this call. + + + + As further notes, the destructors (both + snd_mychip_dev_free and + snd_mychip_free) cannot be defined with + __devexit prefix, because they may be + called from the constructor, too, at the false path. + + + + For a device which allows hotplugging, you can use + snd_card_free_in_thread. This one will + postpone the destruction and wait in a kernel-thread until all + devices are closed. + + +
+ + + + + + + + + PCI Resource Managements + +
+ Full Code Example + + In this section, we'll finish the chip-specific constructor, + destructor and PCI entries. The example code is shown first, + below. + + + PCI Resource Managements Example + +irq >= 0) + free_irq(chip->irq, (void *)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(snd_card_t *card, + struct pci_dev *pci, + mychip_t **rchip) + { + mychip_t *chip; + int err; + static snd_device_ops_t ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + if ((err = pci_enable_device(pci)) < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, 0x0fffffff) < 0 || + pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kcalloc(1, 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 */ + if ((err = pci_request_regions(pci, "My Chip")) < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + SA_INTERRUPT|SA_SHIRQ, "My Chip", + (void *)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) + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, + chip, &ops)) < 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), + }; + + /* initialization of the module */ + static int __init alsa_card_mychip_init(void) + { + return pci_module_init(&driver); + } + + /* clean up the module */ + 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 */ +]]> + + + +
+ +
+ Some Hafta's + + The allocation of PCI resources is done in the + probe() function, and usually an extra + xxx_create() function is written for this + purpose. + + + + In the case of PCI devices, you have to call at first + pci_enable_device() 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 pci_set_master() function, + too. + + + + Suppose the 28bit mask, and the code to be added would be like: + + + + + + + +
+ +
+ Resource Allocation + + The allocation of I/O ports and irqs are 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 ALSA 0.5.x. + + + + Now assume that this PCI device has an I/O port with 8 bytes + and an interrupt. Then mychip_t will have the + following fields: + + + + + + + + + + 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 + kcalloc() automatically, so you + don't have to take care of resetting them. + + + + The allocation of an i/o port is done like this: + + + +port = pci_resource_start(pci, 0); +]]> + + + + + + + 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 kmalloc() by + request_region(). The pointer must be + released via kfree(), but there is some + problem regarding this. This issue will be explained more below. + + + + The allocation of an interrupt source is done like this: + + + +irq, snd_mychip_interrupt, + SA_INTERRUPT|SA_SHIRQ, "My Chip", + (void *)chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; +]]> + + + + where snd_mychip_interrupt() is the + interrupt handler defined later. + Note that chip->irq should be defined + only when request_irq() succeeded. + + + + On the PCI bus, the interrupts can be shared. Thus, + SA_SHIRQ is given as the interrupt flag of + request_irq(). + + + + The last argument of request_irq() 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. + + + + I won't define the detail of the interrupt handler at this + point, but at least its appearance can be explained now. The + interrupt handler looks usually like the following: + + + + + + + + + + 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 is not written here. + + + + For releasing the resources, check-and-release + method is a safer way. For the interrupt, do like this: + + + +irq >= 0) + free_irq(chip->irq, (void *)chip); +]]> + + + + 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. + + + + When you requested I/O ports or memory regions via + pci_request_region() or + pci_request_regions() like this example, + release the resource(s) using the corresponding function, + pci_release_region() or + pci_release_regions(). + + + +pci); +]]> + + + + + + When you requested manually via request_region() + or request_mem_region, you can release it via + release_resource(). Suppose that you keep + the resource pointer returned from request_region() + in chip->res_port, the release procedure looks like below: + + + +res_port) { + release_resource(chip->res_port); + kfree_nocheck(chip->res_port); + } +]]> + + + + As you can see, the resource pointer is also to be freed + via kfree_nocheck() after + release_resource() is called. You + cannot use kfree() here, because on ALSA, + kfree() may be a wrapper to its own + allocator with the memory debugging. Since the resource pointer + is allocated externally outside the ALSA, it must be released + via the native + kfree(). + kfree_nocheck() is used for that; it calls + the native kfree() without wrapper. + + + + Don't forget to call pci_disable_device() + before all finished. + + + + And finally, release the chip-specific record. + + + + + + + + + + Again, remember that you cannot + set __devexit prefix for this destructor. + + + + 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 the hardware-disabling + if the hardware was not initialized yet. + + + + When the chip-data is assigned to the card using + snd_device_new() with + SNDRV_DEV_LOWLELVEL , its destructor is + called at the last. That is, it is assured that all other + components like PCMs and controls have been already released. + You don't have to call stopping PCMs, etc. explicitly, but just + stop the hardware in the low-level. + + + + 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: + + + + + + + + and the allocation would be like below: + + + +iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); +]]> + + + + and the corresponding destructor would be: + + + +iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } +]]> + + + + +
+ +
+ Registration of Device Struct + + At some point, typically after calling snd_device_new(), + you need to register the struct device of the chip + you're handling for udev and co. ALSA provides a macro for compatibility with + older kernels. Simply call like the following: + + +dev); +]]> + + + 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. + + + 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.) + +
+ +
+ PCI Entries + + So far, so good. Let's finish the rest of missing PCI + stuffs. At first, we need a + pci_device_id table for this + chipset. It's a table of PCI vendor/device ID number, and some + masks. + + + + For example, + + + + + + + + + + The first and second fields of + pci_device_id struct are the vendor and + device IDs. If you have nothing special to filter the matching + devices, you can use the rest of fields like above. The last + field of pci_device_id struct is a + private data for this entry. You can specify any value here, for + example, to tell the type of different operations per each + device IDs. Such an example is found in intel8x0 driver. + + + + The last entry of this list is the terminator. You must + specify this all-zero entry. + + + + Then, prepare the pci_driver record: + + + + + + + + + + The probe and + remove functions are what we already + defined in + the previous sections. The remove should + be defined with + __devexit_p() macro, so that it's not + defined for built-in (and non-hot-pluggable) case. The + name + field is the name string of this device. Note that you must not + use a slash / in this string. + + + + And at last, the module entries: + + + + + + + + + + Note that these module entries are tagged with + __init and + __exit prefixes, not + __devinit nor + __devexit. + + + + Oh, one thing was forgotten. If you have no exported symbols, + you need to declare it on 2.2 or 2.4 kernels (on 2.6 kernels + it's not necessary, though). + + + + + + + + That's all! + +
+ + + + + + + + PCM Interface + +
+ General + + 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. + + + + For accessing to the PCM layer, you need to include + <sound/pcm.h> above all. In addition, + <sound/pcm_params.h> might be needed + if you access to some functions related with hw_param. + + + + 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 number. Once when 64bit device number is + used, we'll have more available pcm instances. + + + + A pcm instance consists of pcm playback and capture streams, + and each pcm stream consists of one or more pcm substreams. Some + soundcard supports the multiple-playback function. 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 succeeding open will result in the blocking + or the error with EAGAIN according to the + file open mode. But you don't have to know the detail in your + driver. The PCM middle layer will take all such jobs. + +
+ +
+ Full Code Example + + The example code below does not include any hardware access + routines but shows only the skeleton, how to build up the PCM + interfaces. + + + PCM Example Code + + + .... + + /* hardware definition */ + static snd_pcm_hardware_t 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 snd_pcm_hardware_t 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(snd_pcm_substream_t *substream) + { + mychip_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *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(snd_pcm_substream_t *substream) + { + mychip_t *chip = snd_pcm_substream_chip(substream); + // the hardware-specific codes will be here + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(snd_pcm_substream_t *substream) + { + mychip_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *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(snd_pcm_substream_t *substream) + { + mychip_t *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(snd_pcm_substream_t *substream, + snd_pcm_hw_params_t * hw_params) + { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(snd_pcm_substream_t *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(snd_pcm_substream_t *substream) + { + mychip_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *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_area, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(snd_pcm_substream_t *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(snd_pcm_substream_t *substream) + { + mychip_t *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 snd_pcm_ops_t 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 snd_pcm_ops_t 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(mychip_t *chip) + { + snd_pcm_t *pcm; + int err; + + if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, + &pcm)) < 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; + } +]]> + + + +
+ +
+ Constructor + + A pcm instance is allocated by snd_pcm_new() + function. It would be better to create a constructor for pcm, + namely, + + + +card, "My Chip", 0, 1, 1, + &pcm)) < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } +]]> + + + + + + The snd_pcm_new() function takes the four + arguments. The first argument is the card pointer to which this + pcm is assigned, and the second is the ID string. + + + + The third argument (index, 0 in the + above) is the index of this new pcm. It begins from zero. When + you will create more than one pcm instances, specify the + different numbers in this argument. For example, + index = 1 for the second PCM device. + + + + The fourth and fifth arguments are the number of substreams + for playback and capture, respectively. Here both 1 are given in + the above example. When no playback or no capture is available, + pass 0 to the corresponding argument. + + + + 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 + snd_pcm_substream_t data passed to each callback + as follows: + + + +number; +]]> + + + + + + After the pcm is created, you need to set operators for each + pcm stream. + + + + + + + + + + The operators are defined typically like this: + + + + + + + + Each of callbacks is explained in the subsection + + Operators. + + + + After setting the operators, most likely you'd like to + pre-allocate the buffer. For the pre-allocation, simply call + the following: + + + +pci), + 64*1024, 64*1024); +]]> + + + + It will allocate up to 64kB buffer as default. The details of + buffer management will be described in the later section Buffer and Memory + Management. + + + + Additionally, you can set some extra information for this pcm + in pcm->info_flags. + The available values are defined as + SNDRV_PCM_INFO_XXX in + <sound/asound.h>, which is used for + the hardware definition (described later). When your soundchip + supports only half-duplex, specify like this: + + + +info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; +]]> + + + +
+ +
+ ... And the Destructor? + + 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 destructor + explicitly. + + + + The destructor would be necessary when you created some + special records internally and need to release them. In such a + case, set the destructor function to + pcm->private_free: + + + PCM Instance with a Destructor + +my_private_pcm_data); + // do what you like else + .... + } + + static int __devinit snd_mychip_new_pcm(mychip_t *chip) + { + snd_pcm_t *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } +]]> + + + +
+ +
+ Runtime Pointer - The Chest of PCM Information + + When the PCM substream is opened, a PCM runtime instance is + allocated and assigned to the substream. This pointer is + accessible via substream->runtime. + This runtime pointer holds the various information; it holds + the copy of hw_params and sw_params configurations, the buffer + pointers, mmap records, spinlocks, etc. Almost everyhing you + need for controlling the PCM can be found there. + + + + The definition of runtime instance is found in + <sound/pcm.h>. Here is the + copy from the file. + + + + + + + + + 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 these info. 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 snd_pcm_lib_malloc_pages(), + you don't need to set the DMA buffer information by yourself. + + + + In the sections below, important records are explained. + + +
+ Hardware Description + + The hardware descriptor (snd_pcm_hardware_t) + contains the definitions of the fundamental hardware + configuration. Above all, you'll need to define this in + + the open callback. + 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 + (runtime->hw) 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: + + +runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; +]]> + + + + + + Typically, you'll have a hardware descriptor like below: + + + + + + + + + + + The info field contains the type and + capabilities of this pcm. The bit flags are defined in + <sound/asound.h> as + SNDRV_PCM_INFO_XXX. Here, at least, you + have to specify whether the mmap is supported and which + interleaved format is supported. + When the mmap is supported, add + SNDRV_PCM_INFO_MMAP flag here. When the + hardware supports the interleaved or the non-interleaved + format, SNDRV_PCM_INFO_INTERLEAVED or + SNDRV_PCM_INFO_NONINTERLEAVED flag must + be set, respectively. If both are supported, you can set both, + too. + + + + In the above example, MMAP_VALID and + BLOCK_TRANSFER are specified for OSS mmap + mode. Usually both are set. Of course, + MMAP_VALID is set only if the mmap is + really supported. + + + + The other possible flags are + SNDRV_PCM_INFO_PAUSE and + SNDRV_PCM_INFO_RESUME. The + PAUSE bit means that the pcm supports the + pause operation, while the + RESUME bit means that the pcm supports + the suspend/resume operation. If these flags + are set, the trigger callback below + must handle the corresponding commands. + + + + When the PCM substreams can be synchronized (typically, + synchorinized start/stop of a playback and a capture streams), + you can give SNDRV_PCM_INFO_SYNC_START, + 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. + + + + + + formats field contains the bit-flags + of supported formats (SNDRV_PCM_FMTBIT_XXX). + 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. + + + + + + rates field contains the bit-flags of + supported rates (SNDRV_PCM_RATE_XXX). + When the chip supports continuous rates, pass + CONTINUOUS bit additionally. + The pre-defined rate bits are provided only for typical + rates. If your chip supports unconventional rates, you need to add + KNOT bit and set up the hardware + constraint manually (explained later). + + + + + + rate_min and + rate_max define the minimal and + maximal sample rate. This should correspond somehow to + rates bits. + + + + + + channel_min and + channel_max + define, as you might already expected, the minimal and maximal + number of channels. + + + + + + buffer_bytes_max defines the + maximal buffer size in bytes. There is no + buffer_bytes_min field, since + it can be calculated from the minimal period size and the + minimal number of periods. + Meanwhile, period_bytes_min and + define the minimal and maximal size of the period in bytes. + periods_max and + periods_min define the maximal and + minimal number of periods in the buffer. + + + + The period is a term, that corresponds to + fragment in the OSS world. The period defines the size at + which the 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. + + + + + + There is also a field fifo_size. + This specifies the size of the hardware FIFO, but it's not + used currently in the driver nor in the alsa-lib. So, you + can ignore this field. + + + + +
+ +
+ PCM Configurations + + 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 on runtime instance + after the application sends hw_params data via + alsa-lib. There are many fields copied from hw_params and + sw_params structs. For example, + format holds the format type + chosen by the application. This field contains the enum value + SNDRV_PCM_FORMAT_XXX. + + + + One thing to be noted is that the configured buffer and period + sizes are stored in frames in the runtime + In the ALSA world, 1 frame = channels * samples-size. + For conversion between frames and bytes, you can use the + helper functions, frames_to_bytes() and + bytes_to_frames(). + + +period_size); +]]> + + + + + + Also, many software parameters (sw_params) are + stored in frames, too. Please check the type of the field. + snd_pcm_uframes_t is for the frames as unsigned + integer while snd_pcm_sframes_t is for the frames + as signed integer. + +
+ +
+ DMA Buffer Information + + The DMA buffer is defined by the following four fields, + dma_area, + dma_addr, + dma_bytes and + dma_private. + The dma_area holds the buffer + pointer (the logical address). You can call + memcpy from/to + this pointer. Meanwhile, dma_addr + holds the physical address of the buffer. This field is + specified only when the buffer is a linear buffer. + dma_bytes holds the size of buffer + in bytes. dma_private is used for + the ALSA DMA allocator. + + + + If you use a standard ALSA function, + snd_pcm_lib_malloc_pages(), for + allocating the buffer, these fields are set by the ALSA middle + layer, and you should not 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, dma_bytes is mandatory. + dma_area is necessary when the + buffer is mmapped. If your driver doesn't support mmap, this + field is not necessary. dma_addr + is also not mandatory. You can use + dma_private as you like, too. + +
+ +
+ Running Status + + The running status can be referred via runtime->status. + This is the pointer to snd_pcm_mmap_status_t + record. For example, you can get the current DMA hardware + pointer via runtime->status->hw_ptr. + + + + The DMA application pointer can be referred via + runtime->control, which points + snd_pcm_mmap_control_t record. + However, accessing directly to this value is not recommended. + +
+ +
+ Private Data + + You can allocate a record for the substream and store it in + runtime->private_data. Usually, this + done in + + the open callback. + Don't mix this with pcm->private_data. + The pcm->private_data usually points the + chip instance assigned statically at the creation of PCM, while the + runtime->private_data points a dynamic + data created at the PCM open callback. + + + +runtime->private_data = data; + .... + } +]]> + + + + + + The allocated object must be released in + + the close callback. + +
+ +
+ Interrupt Callbacks + + The field transfer_ack_begin and + transfer_ack_end are called at + the beginning and the end of + snd_pcm_period_elapsed(), respectively. + +
+ +
+ +
+ Operators + + OK, now let me explain the detail of each pcm callback + (ops). In general, every callback must + return 0 if successful, or a negative number with the error + number such as -EINVAL at any + error. + + + + The callback function takes at least the argument with + snd_pcm_substream_t pointer. For retrieving the + chip record from the given substream instance, you can use the + following macro. + + + + + + + + The macro reads substream->private_data, + which is a copy of pcm->private_data. + You can override the former if you need to assign different data + records per PCM substream. For example, cmi8330 driver assigns + different private_data for playback and capture directions, + because it uses two different codecs (SB- and AD-compatible) for + different directions. + + +
+ open callback + + + + + + + + This is called when a pcm substream is opened. + + + + At least, here you have to initialize the runtime->hw + record. Typically, this is done by like this: + + + +runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } +]]> + + + + where snd_mychip_playback_hw is the + pre-defined hardware description. + + + + You can allocate a private data in this callback, as described + in + Private Data section. + + + + If the hardware configuration needs more constraints, set the + hardware constraints here, too. + See + Constraints for more details. + +
+ +
+ close callback + + + + + + + + Obviously, this is called when a pcm substream is closed. + + + + Any private instance for a pcm substream allocated in the + open callback will be released here. + + + +runtime->private_data); + .... + } +]]> + + + +
+ +
+ ioctl callback + + This is used for any special action to pcm ioctls. But + usually you can pass a generic ioctl callback, + snd_pcm_lib_ioctl. + +
+ +
+ hw_params callback + + + + + + + + This and hw_free callbacks exist + only on ALSA 0.9.x. + + + + This is called when the hardware parameter + (hw_params) 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. + + + + Many hardware set-up should be done in this callback, + including the allocation of buffers. + + + + Parameters to be initialized are retrieved by + params_xxx() macros. For allocating a + buffer, you can call a helper function, + + + + + + + + snd_pcm_lib_malloc_pages() is available + only when the DMA buffers have been pre-allocated. + See the section + Buffer Types for more details. + + + + Note that this and prepare callbacks + may be called multiple times per initialization. + For example, the OSS emulation may + call these callbacks at each change via its ioctl. + + + + Thus, you need to take care not to allocate the same buffers + many times, which will lead to memory leak! Calling the + helper function above many times is OK. It will release the + previous buffer automatically when it was already allocated. + + + + Another note is that this callback is non-atomic + (schedulable). This is important, because the + trigger callback + is atomic (non-schedulable). That is, mutex or any + schedule-related functions are not available in + trigger callback. + Please see the subsection + + Atomicity for details. + +
+ +
+ hw_free callback + + + + + + + + + + This is called to release the resources allocated via + hw_params. For example, releasing the + buffer via + snd_pcm_lib_malloc_pages() is done by + calling the following: + + + + + + + + + + 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. + +
+ +
+ prepare callback + + + + + + + + + + This callback is called when the pcm is + prepared. You can set the format type, sample + rate, etc. here. The difference from + hw_params is that the + prepare callback will be called at each + time + snd_pcm_prepare() is called, i.e. when + recovered after underruns, etc. + + + + Note that this callback became non-atomic since the recent version. + You can use schedule-related fucntions safely in this callback now. + + + + 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. + + + + Be careful that this callback will be called many times at + each set up, too. + +
+ +
+ trigger callback + + + + + + + + This is called when the pcm is started, stopped or paused. + + + + Which action is specified in the second argument, + SNDRV_PCM_TRIGGER_XXX in + <sound/pcm.h>. At least, + START and STOP + commands must be defined in this callback. + + + + + + + + + + When the pcm supports the pause operation (given in info + field of the hardware table), PAUSE_PUSE + and PAUSE_RELEASE commands must be + handled here, too. The former is the command to pause the pcm, + and the latter to restart the pcm again. + + + + When the pcm supports the suspend/resume operation + (i.e. SNDRV_PCM_INFO_RESUME flag is set), + SUSPEND and RESUME + commands must be handled, too. + These commands are issued when the power-management status is + changed. Obviously, the SUSPEND and + RESUME + do suspend and resume of the pcm substream, and usually, they + are identical with STOP and + START commands, respectively. + + + + As mentioned, this callback is atomic. You cannot call + the function going to 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. + +
+ +
+ pointer callback + + + + + + + + This callback is called when the PCM middle layer inquires + the current hardware position on the buffer. The position must + be returned in frames (which was in bytes on ALSA 0.5.x), + ranged from 0 to buffer_size - 1. + + + + This is called usually from the buffer-update routine in the + pcm middle layer, which is invoked when + snd_pcm_period_elapsed() 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. + + + + This callback is also atomic. + +
+ +
+ copy and silence callbacks + + These callbacks are not mandatory, and can be omitted in + most cases. These callbacks are used when the hardware buffer + cannot be on 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. + + + + If these two callbacks are defined, copy and set-silence + operations are done by them. The detailed will be described in + the later section Buffer and Memory + Management. + +
+ +
+ ack callback + + 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. + + + This callback is atomic. + +
+ +
+ page callback + + + This callback is also not mandatory. This callback is used + mainly for the non-contiguous buffer. The mmap calls this + callback to get the page address. Some examples will be + explained in the later section Buffer and Memory + Management, too. + +
+
+ +
+ Interrupt Handler + + 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 snd_pcm_period_elapsed() + function. + + + + There are several types of sound chips to generate the interrupts. + + +
+ Interrupts at the period (fragment) boundary + + This is the most frequently found type: the hardware + generates an interrupt at each period boundary. + In this case, you can call + snd_pcm_period_elapsed() at each + interrupt. + + + + snd_pcm_period_elapsed() 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). + + + + If you aquire 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 + snd_pcm_period_elapsed(), because + snd_pcm_period_elapsed() calls other pcm + callbacks inside. + + + + A typical coding would be like: + + + Interrupt Handler Case #1 + +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; + } +]]> + + + +
+ +
+ High-frequent timer interrupts + + This is the case when the hardware doesn't generate interrupts + at the period boundary but do timer-interrupts at the fixed + timer rate (e.g. es1968 or ymfpci drivers). + In this case, you need to check the current hardware + position and accumulates the processed sample length at each + interrupt. When the accumulated size overcomes the period + size, call + snd_pcm_period_elapsed() and reset the + accumulator. + + + + A typical coding would be like the following. + + + Interrupt Handler Case #2 + +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; + } +]]> + + + +
+ +
+ On calling <function>snd_pcm_period_elapsed()</function> + + In both cases, even if more than one period are elapsed, you + don't have to call + snd_pcm_period_elapsed() many times. Call + only once. And the pcm layer will check the current hardware + pointer and update to the latest status. + +
+
+ +
+ Atomicity + + One of the most important (and thus difficult to debug) problem + on the kernel programming is the race condition. + On linux kernel, usually it's solved via spin-locks or + semaphores. In general, if the race condition may + happen in the interrupt handler, it's handled as atomic, and you + have to use spinlock for protecting the critical session. If it + never happens in the interrupt and it may take relatively long + time, you should use semaphore. + + + + As already seen, some pcm callbacks are atomic and some are + not. For example, hw_params callback is + non-atomic, while trigger 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 use a spinlock or a semaphore in the callbacks. + + + + In the atomic callbacks, you cannot use functions which may call + schedule or go to + sleep. The semaphore and mutex do sleep, + and hence they cannot be used inside the atomic callbacks + (e.g. trigger callback). + For taking a certain delay in such a callback, please use + udelay() or mdelay(). + + + + All three atomic callbacks (trigger, pointer, and ack) are + called with local interrupts disabled. + + +
+
+ Constraints + + If your chip supports unconventional sample rates, or only the + limited samples, you need to set a constraint for the + condition. + + + + For example, in order to restrict the sample rates in the some + supported values, use + snd_pcm_hw_constraint_list(). + You need to call this function in the open callback. + + + Example of Hardware Constraints + +runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } +]]> + + + + + + There are many different constraints. + Look in sound/pcm.h 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 snd_pcm_hardware_t stucture (or in any + other constraint_list). You can build a rule like this: + + + Example of Hardware Constraints for Channels + +min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } +]]> + + + + + + Then you need to call this function to add your rule: + + + +runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, + -1); +]]> + + + + + + 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 of Hardware Constraints for Channels + +bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } +]]> + + + + + + ...and in the open callback: + + +runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); +]]> + + + + + + I won't explain more details here, rather I + would like to say, Luke, use the source. + +
+ + + + + + + + + Control Interface + +
+ General + + The control interface is used widely for many switches, + sliders, etc. which are accessed from the user-space. Its most + important use is the mixer interface. In other words, on ALSA + 0.9.x, all the mixer stuff is implemented on the control kernel + API (while there was an independent mixer kernel API on 0.5.x). + + + + ALSA has a well-defined AC97 control module. If your chip + supports only the AC97 and nothing else, you can skip this + section. + + + + The control API is defined in + <sound/control.h>. + Include this file if you add your own controls. + +
+ +
+ Definition of Controls + + For creating a new control, you need to define the three + callbacks: info, + get and + put. Then, define a + snd_kcontrol_new_t record, such as: + + + Definition of a Control + + + + + + + + Most likely the control is created via + snd_ctl_new1(), and in such a case, you can + add __devinitdata prefix to the + definition like above. + + + + The iface field specifies the type of + the control, + SNDRV_CTL_ELEM_IFACE_XXX. There are + MIXER, PCM, + CARD, etc. + + + + The name is the name identifier + string. On 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 subsection + + Control Names. + + + + The index 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. + + + + The access field contains the access + type of this control. Give the combination of bit masks, + SNDRV_CTL_ELEM_ACCESS_XXX, there. + The detailed will be explained in the subsection + + Access Flags. + + + + The private_values field contains + an arbitrary long integer value for this record. When using + generic info, + get and + put 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. + + + + The other three are + + callback functions. + +
+ +
+ Control Names + + There are some standards for defining the control names. A + control is usually defined from the three parts as + SOURCE DIRECTION FUNCTION. + + + + The first, SOURCE, specifies the source + of the control, and is a string such as Master, + PCM, CD or + Line. There are many pre-defined sources. + + + + The second, DIRECTION, is one of the + following strings according to the direction of the control: + Playback, Capture, Bypass + Playback and Bypass Capture. Or, it can + be omitted, meaning both playback and capture directions. + + + + The third, FUNCTION, is one of the + following strings according to the function of the control: + Switch, Volume and + Route. + + + + The example of control names are, thus, Master Capture + Switch or PCM Playback Volume. + + + + There are some exceptions: + + +
+ Global capture and playback + + Capture Source, Capture Switch + and Capture Volume are used for the global + capture (input) source, switch and volume. Similarly, + Playback Switch and Playback + Volume are used for the global output gain switch and + volume. + +
+ +
+ Tone-controls + + tone-control switch and volumes are specified like + Tone Control - XXX, e.g. Tone Control - + Switch, Tone Control - Bass, + Tone Control - Center. + +
+ +
+ 3D controls + + 3D-control switches and volumes are specified like 3D + Control - XXX, e.g. 3D Control - + Switch, 3D Control - Center, 3D + Control - Space. + +
+ +
+ Mic boost + + Mic-boost switch is set as Mic Boost or + Mic Boost (6dB). + + + + More precise information can be found in + Documentation/sound/alsa/ControlNames.txt. + +
+
+ +
+ Access Flags + + + The access flag is the bit-flags which specifies the access type + of the given control. The default access type is + SNDRV_CTL_ELEM_ACCESS_READWRITE, + which means both read and write are allowed to this control. + When the access flag is omitted (i.e. = 0), it is + regarded as READWRITE access as default. + + + + When the control is read-only, pass + SNDRV_CTL_ELEM_ACCESS_READ instead. + In this case, you don't have to define + put callback. + Similarly, when the control is write-only (although it's a rare + case), you can use WRITE flag instead, and + you don't need get callback. + + + + If the control value changes frequently (e.g. the VU meter), + VOLATILE flag should be given. This means + that the control may be changed without + + notification. Applications should poll such + a control constantly. + + + + When the control is inactive, set + INACTIVE flag, too. + There are LOCK and + OWNER flags for changing the write + permissions. + + +
+ +
+ Callbacks + +
+ info callback + + The info callback is used to get + the detailed information of this control. This must store the + values of the given snd_ctl_elem_info_t + object. For example, for a boolean control with a single + element will be: + + + Example of info callback + +type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } +]]> + + + + + + The type field specifies the type + of the control. There are BOOLEAN, + INTEGER, ENUMERATED, + BYTES, IEC958 and + INTEGER64. The + count field specifies the + number of elements in this control. For example, a stereo + volume would have count = 2. The + value field is a union, and + the values stored are depending on the type. The boolean and + integer are identical. + + + + The enumerated type is a bit different from others. You'll + need to set the string for the currently given item index. + + + +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; + } +]]> + + + +
+ +
+ get callback + + + This callback is used to read the current value of the + control and to return to the user-space. + + + + For example, + + + Example of get callback + +value.integer.value[0] = get_some_value(chip); + return 0; + } +]]> + + + + + + Here, the chip instance is retrieved via + snd_kcontrol_chip() macro. This macro + converts from kcontrol->private_data to the type defined by + chip_t. The + kcontrol->private_data field is + given as the argument of snd_ctl_new() + (see the later subsection + Constructor). + + + + The value field is depending on + the type of control as well as on info callback. For example, + the sb driver uses this field to store the register offset, + the bit-shift and the bit-mask. The + private_value is set like + + + + + + and is retrieved in callbacks like + + +private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } +]]> + + + + + + In get callback, you have to fill all the elements if the + control has more than one elements, + i.e. count > 1. + In the example above, we filled only one element + (value.integer.value[0]) since it's + assumed as count = 1. + +
+ +
+ put callback + + + This callback is used to write a value from the user-space. + + + + For example, + + + Example of put callback + +current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } +]]> + + + + 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. + + + + Like get callback, + when the control has more than one elements, + all elemehts must be evaluated in this callback, too. + +
+ +
+ Callbacks are not atomic + + All these three callbacks are basically not atomic. + +
+
+ +
+ Constructor + + When everything is ready, finally we can create a new + control. For creating a control, there are two functions to be + called, snd_ctl_new1() and + snd_ctl_add(). + + + + In the simplest way, you can do like this: + + + + + + + + where my_control is the + snd_kcontrol_new_t object defined above, and chip + is the object pointer to be passed to + kcontrol->private_data + which can be referred in callbacks. + + + + snd_ctl_new1() allocates a new + snd_kcontrol_t instance (that's why the definition + of my_control can be with + __devinitdata + prefix), and snd_ctl_add assigns the given + control component to the card. + +
+ +
+ Change Notification + + If you need to change and update a control in the interrupt + routine, you can call snd_ctl_notify(). For + example, + + + + + + + + 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 snd_ctl_elem_id_t + to be notified. + You can find some examples in es1938.c or + es1968.c for hardware volume interrupts. + +
+ + + + + + + + + API for AC97 Codec + +
+ General + + The ALSA AC97 codec layer is a well-defined one, and you don't + have to write many codes to control it. Only low-level control + routines are necessary. The AC97 codec API is defined in + <sound/ac97_codec.h>. + +
+ +
+ Full Code Example + + + Example of AC97 Interface + +private_data; + .... + // read a register value here from the codec + return the_register_value; + } + + static void snd_mychip_ac97_write(ac97_t *ac97, + unsigned short reg, unsigned short val) + { + mychip_t *chip = ac97->private_data; + .... + // write the given register value to the codec + } + + static int snd_mychip_ac97(mychip_t *chip) + { + ac97_bus_t *bus; + ac97_template_t ac97; + int err; + static ac97_bus_ops_t ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + +]]> + + + +
+ +
+ Constructor + + For creating an ac97 instance, first call snd_ac97_bus + with an ac97_bus_ops_t record with callback functions. + + + + + + + + The bus record is shared among all belonging ac97 instances. + + + + And then call snd_ac97_mixer() with an ac97_template_t + record together with the bus pointer created above. + + + +ac97); +]]> + + + + where chip->ac97 is the pointer of a newly created + ac97_t 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. When 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. + +
+ +
+ Callbacks + + The standard callbacks are read and + write. Obviously they + correspond to the functions for read and write accesses to the + hardware low-level codes. + + + + The read callback returns the + register value specified in the argument. + + + +private_data; + .... + return the_register_value; + } +]]> + + + + Here, the chip can be cast from ac97->private_data. + + + + Meanwhile, the write callback is + used to set the register value. + + + + + + + + + + These callbacks are non-atomic like the callbacks of control API. + + + + There are also other callbacks: + reset, + wait and + init. + + + + The reset callback is used to reset + the codec. If the chip requires a special way of reset, you can + define this callback. + + + + The wait callback is used for a + certain wait at the standard initialization of the codec. If the + chip requires the extra wait-time, define this callback. + + + + The init callback is used for + additional initialization of the codec. + +
+ +
+ Updating Registers in The Driver + + If you need to access to the codec from the driver, you can + call the following functions: + snd_ac97_write(), + snd_ac97_read(), + snd_ac97_update() and + snd_ac97_update_bits(). + + + + Both snd_ac97_write() and + snd_ac97_update() functions are used to + set a value to the given register + (AC97_XXX). The difference between them is + that snd_ac97_update() doesn't write a + value if the given value has been already set, while + snd_ac97_write() always rewrites the + value. + + + + + + + + + + snd_ac97_read() is used to read the value + of the given register. For example, + + + + + + + + + + snd_ac97_update_bits() is used to update + some bits of the given register. + + + + + + + + + + Also, there is a function to change the sample rate (of a + certain register such as + AC97_PCM_FRONT_DAC_RATE) when VRA or + DRA is supported by the codec: + snd_ac97_set_rate(). + + + + + + + + + + The following registers are available for setting the rate: + AC97_PCM_MIC_ADC_RATE, + AC97_PCM_FRONT_DAC_RATE, + AC97_PCM_LR_ADC_RATE, + AC97_SPDIF. When the + AC97_SPDIF is specified, the register is + not really changed but the corresponding IEC958 status bits will + be updated. + +
+ +
+ Clock Adjustment + + On some chip, 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 the auto-measurement function of the + clock. + +
+ +
+ Proc Files + + The ALSA AC97 interface will create a proc file such as + /proc/asound/card0/codec97#0/ac97#0-0 and + ac97#0-0+regs. You can refer to these files to + see the current status and registers of the codec. + +
+ +
+ Multiple Codecs + + When there are several codecs on the same card, you need to + call snd_ac97_new() multiple times with + ac97.num=1 or greater. The num field + specifies the codec + number. + + + + If you have set up multiple codecs, you need to either write + different callbacks for each codec or check + ac97->num in the + callback routines. + +
+ + + + + + + + + MIDI (MPU401-UART) Interface + +
+ General + + 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 + <sound/mpu401.h>. + + + + Some soundchips have similar but a little bit different + implementation of mpu401 stuff. For example, emu10k1 has its own + mpu401 routines. + +
+ +
+ Constructor + + For creating a rawmidi object, call + snd_mpu401_uart_new(). + + + + + + + + + + The first argument is the card pointer, and the second is the + index of this component. You can create up to 8 rawmidi + devices. + + + + The third argument is the type of the hardware, + MPU401_HW_XXX. If it's not a special one, + you can use MPU401_HW_MPU401. + + + + The 4th argument is the i/o port address. Many + backward-compatible MPU401 has 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. + + + + When the i/o port address above is a 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 non-zero + to the 5th argument + (integrated). Otherwise, pass 0 to it, + and + the mpu401-uart layer will allocate the i/o ports by itself. + + + + Usually, the port address corresponds to the command port and + port + 1 corresponds to the data port. If not, you may change + the cport field of + mpu401_t manually + afterward. However, mpu401_t pointer is not + returned explicitly by + snd_mpu401_uart_new(). You need to cast + rmidi->private_data to + mpu401_t explicitly, + + + +private_data; +]]> + + + + and reset the cport as you like: + + + +cport = my_own_control_port; +]]> + + + + + + The 6th argument specifies the irq number for UART. If the irq + is already allocated, pass 0 to the 7th argument + (irq_flags). Otherwise, pass the flags + for irq allocation + (SA_XXX bits) to it, and the irq will be + reserved by the mpu401-uart layer. If the card doesn't generates + UART interrupts, pass -1 as the irq number. Then a timer + interrupt will be invoked for polling. + +
+ +
+ Interrupt Handler + + When the interrupt is allocated in + snd_mpu401_uart_new(), the private + interrupt handler is used, hence you don't have to do nothing + else than creating the mpu401 stuff. Otherwise, you have to call + snd_mpu401_uart_interrupt() explicitly when + a UART interrupt is invoked and checked in your own interrupt + handler. + + + + In this case, you need to pass the private_data of the + returned rawmidi object from + snd_mpu401_uart_new() as the second + argument of snd_mpu401_uart_interrupt(). + + + +private_data, regs); +]]> + + + +
+ + + + + + + + + RawMIDI Interface + +
+ Overview + + + 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. + + + + 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. + + + + The rawmidi API is defined in + <sound/rawmidi.h>. + +
+ +
+ Constructor + + + To create a rawmidi device, call the + snd_rawmidi_new function: + + +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; +]]> + + + + + + The first argument is the card pointer, the second argument is + the ID string. + + + + The third argument is the index of this component. You can + create up to 8 rawmidi devices. + + + + 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.) + + + + Set the info_flags field to specify + the capabilities of the device. + Set SNDRV_RAWMIDI_INFO_OUTPUT if there is + at least one output port, + SNDRV_RAWMIDI_INFO_INPUT if there is at + least one input port, + and SNDRV_RAWMIDI_INFO_DUPLEX if the device + can handle output and input at the same time. + + + + 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 substream of a device: + + + + + + + + + The operators are usually defined like this: + + + + + + These callbacks are explained in the Callbacks + section. + + + + If there is more than one substream, you should give each one a + unique name: + + +streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { + substream = list_entry(list, snd_rawmidi_substream_t, list); + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ +]]> + + + +
+ +
+ Callbacks + + + In all callbacks, the private data that you've set for the + rawmidi device can be accessed as + substream->rmidi->private_data. + + + + + If there is more than one port, your callbacks can determine the + port index from the snd_rawmidi_substream_t data passed to each + callback: + + +number; +]]> + + + + +
+ <function>open</function> callback + + + + + + + + + This is called when a substream is opened. + You can initialize the hardware here, but you should not yet + start transmitting/receiving data. + +
+ +
+ <function>close</function> callback + + + + + + + + + Guess what. + + + + The open and close + callbacks of a rawmidi device are serialized with a mutex, + and can sleep. + +
+ +
+ <function>trigger</function> callback for output + substreams + + + + + + + + + This is called with a nonzero up + parameter when there is some data in the substream buffer that + must be transmitted. + + + + To read data from the buffer, call + snd_rawmidi_transmit_peek. It will + return the number of bytes that have been read; this will be + less than the number of bytes requested when there is no more + data in the buffer. + After the data has been transmitted successfully, call + snd_rawmidi_transmit_ack to remove the + data from the substream buffer: + + + + + + + + + If you know beforehand that the hardware will accept data, you + can use the snd_rawmidi_transmit function + which reads some data and removes it from the buffer at once: + + + + + + + + + If you know beforehand how many bytes you can accept, you can + use a buffer size greater than one with the + snd_rawmidi_transmit* functions. + + + + The trigger 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. + + + + The trigger callback is called with a + zero up parameter when the transmission + of data should be aborted. + +
+ +
+ <function>trigger</function> callback for input + substreams + + + + + + + + + This is called with a nonzero up + parameter to enable receiving data, or with a zero + up parameter do disable receiving data. + + + + The trigger callback must not sleep; the + actual reading of data from the device is usually done in an + interrupt handler. + + + + When data reception is enabled, your interrupt handler should + call snd_rawmidi_receive for all received + data: + + + + + + +
+ +
+ <function>drain</function> callback + + + + + + + + + This is only used with output substreams. This function should wait + until all data read from the substream buffer has been transmitted. + This ensures that the device can be closed and the driver unloaded + without losing data. + + + + This callback is optional. If you do not set + drain in the snd_rawmidi_ops_t + structure, ALSA will simply wait for 50 milliseconds + instead. + +
+
+ + + + + + + + + Miscellaneous Devices + +
+ FM OPL3 + + The FM OPL3 is still used on many chips (mainly for backward + compatibility). ALSA has a nice OPL3 FM control layer, too. The + OPL3 API is defined in + <sound/opl3.h>. + + + + FM registers can be directly accessed through direct-FM API, + defined in <sound/asound_fm.h>. In + ALSA native mode, FM registers are accessed through + Hardware-Dependant Device direct-FM extension API, whereas in + OSS compatible mode, FM registers can be accessed with OSS + direct-FM compatible API on /dev/dmfmX device. + + + + For creating the OPL3 component, you have two functions to + call. The first one is a constructor for opl3_t + instance. + + + + + + + + + + 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. + + + + The fourth argument is the hardware type. + + + + When the left and right ports have been already allocated by + the card driver, pass non-zero to the fifth argument + (integrated). Otherwise, opl3 module will + allocate the specified ports by itself. + + + + When the accessing to the hardware requires special method + instead of the standard I/O access, you can create opl3 instance + separately with snd_opl3_new(). + + + + + + + + + + Then set command, + private_data and + private_free 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 opl3->private_data field. + + + + After creating the opl3 instance via snd_opl3_new(), + call snd_opl3_init() to initialize the chip to the + proper state. Note that snd_opl3_create() always + calls it internally. + + + + If the opl3 instance is created successfully, then create a + hwdep device for this opl3. + + + + + + + + + + The first argument is the opl3_t instance you + created, and the second is the index number, usually 0. + + + + 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). + +
+ +
+ Hardware-Dependent Devices + + Some chips need the access from the user-space 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 <sound/hwdep.h>. You can + find examples in opl3 driver or + isa/sb/sb16_csp.c. + + + + Creation of the hwdep instance is done via + snd_hwdep_new(). + + + + + + + + where the third argument is the index number. + + + + You can then pass any pointer value to the + private_data. + If you assign a private data, you should define the + destructor, too. The destructor function is set to + private_free field. + + + +private_data = p; + hw->private_free = mydata_free; +]]> + + + + and the implementation of destructor would be: + + + +private_data; + kfree(p); + } +]]> + + + + + + The arbitrary file operations can be defined for this + instance. The file operators are defined in + ops table. For example, assume that + this chip needs an ioctl. + + + +ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; +]]> + + + + And implement the callback functions as you like. + +
+ +
+ IEC958 (S/PDIF) + + Usually the controls for IEC958 devices are implemented via + control interface. There is a macro to compose a name string for + IEC958 controls, SNDRV_CTL_NAME_IEC958() + defined in <include/asound.h>. + + + + There are some standard controls for IEC958 status bits. These + controls use the type SNDRV_CTL_ELEM_TYPE_IEC958, + and the size of element is fixed as 4 bytes array + (value.iec958.status[x]). For info + callback, you don't specify + the value field for this type (the count field must be set, + though). + + + + IEC958 Playback Con Mask is used to return the + bit-mask for the IEC958 status bits of consumer mode. Similarly, + IEC958 Playback Pro Mask returns the bitmask for + professional mode. They are read-only controls, and are defined + as MIXER controls (iface = + SNDRV_CTL_ELEM_IFACE_MIXER). + + + + Meanwhile, IEC958 Playback Default control is + defined for getting and setting the current default IEC958 + bits. Note that this one is usually defined as a PCM control + (iface = SNDRV_CTL_ELEM_IFACE_PCM), + although in some places it's defined as a MIXER control. + + + + 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 + IEC958 xxx, preferably using + SNDRV_CTL_NAME_IEC958() macro. + + + + You can find several cases, for example, + pci/emu10k1, + pci/ice1712, or + pci/cmipci.c. + +
+ + + + + + + + + Buffer and Memory Management + +
+ Buffer Types + + 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 + snd_malloc_xxx_pages() function, where xxx + is the bus type. + + + + The allocation of pages with fallback is + snd_malloc_xxx_pages_fallback(). This + function tries to allocate the specified pages but if the pages + are not available, it tries to reduce the page sizes until the + enough space is found. + + + + For releasing the space, call + snd_free_xxx_pages() function. + + + + 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 pre-allocation. + As already written, you can call the following function at the + construction of pcm instance (in the case of PCI bus). + + + + + + + + where size is the byte size to be + pre-allocated and the max is the maximal + size to be changed via prealloc proc file. + The allocator will try to get as large area as possible + within the given size. + + + + The second argument (type) and the third argument (device pointer) + are dependent on the bus. + In the case of ISA bus, pass snd_dma_isa_data() + as the third argument with SNDRV_DMA_TYPE_DEV type. + For the continuous buffer unrelated to the bus can be pre-allocated + with SNDRV_DMA_TYPE_CONTINUOUS type and the + snd_dma_continuous_data(GFP_KERNEL) device pointer, + whereh GFP_KERNEL is the kernel allocation flag to + use. For the SBUS, SNDRV_DMA_TYPE_SBUS and + snd_dma_sbus_data(sbus_dev) are used instead. + For the PCI scatter-gather buffers, use + SNDRV_DMA_TYPE_DEV_SG with + snd_dma_pci_data(pci) + (see the section + Non-Contiguous Buffers + ). + + + + Once when the buffer is pre-allocated, you can use the + allocator in the hw_params callback + + + + + + + + Note that you have to pre-allocate to use this function. + +
+ +
+ External Hardware Buffers + + 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). + + + + The first case works fine if the external hardware buffer is enough + large. This method doesn't need any extra buffers and thus is + more effective. You need to define the + copy and + silence 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. + + + + The second case allows the mmap of the buffer, although you have + to handle an interrupt or a tasklet for transferring the data + from the intermediate buffer to the hardware buffer. You can find an + example in vxpocket driver. + + + + Another case is that 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 intel. In + non-mmap mode, the data cannot be transferred as the normal + way. Thus you need to define copy and + silence callbacks as well + as in the cases above. The examples are found in + rme32.c and rme96.c. + + + + The implementation of copy and + silence callbacks depends upon + whether the hardware supports interleaved or non-interleaved + samples. The copy callback is + defined like below, a bit + differently depending whether the direction is playback or + capture: + + + + + + + + + + In the case of interleaved samples, the second argument + (channel) is not used. The third argument + (pos) points the + current position offset in frames. + + + + 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. + + + + The last argument is the number of frames to be copied. + + + + What you have to do in this callback is again different + between playback and capture directions. In the case of + playback, you do: copy the given amount of data + (count) at the specified pointer + (src) to the specified offset + (pos) on the hardware buffer. When + coded like memcpy-like way, the copy would be like: + + + + + + + + + + For the capture direction, you do: copy the given amount of + data (count) at the specified offset + (pos) on the hardware buffer to the + specified pointer (dst). + + + + + + + + Note that both of the position and the data amount are given + in frames. + + + + In the case of non-interleaved samples, the implementation + will be a bit more complicated. + + + + 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 + isa/gus/gus_pcm.c as an example. + + + + The silence callback is also + implemented in a similar way. + + + + + + + + + + The meanings of arguments are identical with the + copy + callback, although there is no src/dst + argument. In the case of interleaved samples, the channel + argument has no meaning, as well as on + copy callback. + + + + The role of silence callback is to + set the given amount + (count) of silence data at the + specified offset (pos) 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: + + + + + + + + + + In the case of non-interleaved samples, again, the + implementation becomes a bit more complicated. See, for example, + isa/gus/gus_pcm.c. + +
+ +
+ Non-Contiguous Buffers + + If your hardware supports the page table like emu10k1 or the + buffer descriptors like via82xx, you can use the scatter-gather + (SG) DMA. ALSA provides an interface for handling SG-buffers. + The API is provided in <sound/pcm.h>. + + + + For creating the SG-buffer handler, call + snd_pcm_lib_preallocate_pages() or + snd_pcm_lib_preallocate_pages_for_all() + with SNDRV_DMA_TYPE_DEV_SG + in the PCM constructor like other PCI pre-allocator. + You need to pass the snd_dma_pci_data(pci), + where pci is the struct pci_dev pointer + of the chip as well. + The snd_sg_buf_t instance is created as + substream->dma_private. You can cast + the pointer like: + + + +dma_private; +]]> + + + + + + Then call snd_pcm_lib_malloc_pages() + in hw_params 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 + snd_pcm_sgbuf_get_addr(). + + + + When a SG-handler is used, you need to set + snd_pcm_sgbuf_ops_page as + the page callback. + (See + page callback section.) + + + + For releasing the data, call + snd_pcm_lib_free_pages() in the + hw_free callback as usual. + +
+ +
+ Vmalloc'ed Buffers + + It's possible to use a buffer allocated via + vmalloc, for example, for an intermediate + buffer. Since the allocated pages are not contiguous, you need + to set the page callback to obtain + the physical address at every offset. + + + + The implementation of page callback + would be like this: + + + + + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(snd_pcm_substream_t *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } +]]> + + + +
+ + + + + + + + + Proc Interface + + 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 + <sound/info.h>. + + + + For creating a proc file, call + snd_card_proc_new(). + + + + + + + + where the second argument specifies the proc-file name to be + created. The above example will create a file + my-file under the card directory, + e.g. /proc/asound/card0/my-file. + + + + Like other components, the proc entry created via + snd_card_proc_new() will be registered and + released automatically in the card registration and release + functions. + + + + When the creation is successful, the function stores a new + instance at the pointer given in the third argument. + It is initialized as a text proc file for read only. For using + this proc file as a read-only text file as it is, set the read + callback with a private data via + snd_info_set_text_ops(). + + + + + + + + where the second argument (chip) is the + private data to be used in the callbacks. The third parameter + specifies the read buffer size and the fourth + (my_proc_read) is the callback function, which + is defined like + + + + + + + + + + + In the read callback, use snd_iprintf() for + output strings, which works just like normal + printf(). For example, + + + +private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } +]]> + + + + + + The file permission can be changed afterwards. As default, it's + set as read only for all users. If you want to add the write + permission to the user (root as default), set like below: + + + +mode = S_IFREG | S_IRUGO | S_IWUSR; +]]> + + + + and set the write buffer size and the callback + + + +c.text.write_size = 256; + entry->c.text.write = my_proc_write; +]]> + + + + + + The buffer size for read is set to 1024 implicitly by + snd_info_set_text_ops(). It should suffice + in most cases (the size will be aligned to + PAGE_SIZE anyway), but if you need to handle + very large text files, you can set it explicitly, too. + + + +c.text.read_size = 65536; +]]> + + + + + + For the write callback, you can use + snd_info_get_line() to get a text line, and + snd_info_get_str() to retrieve a string from + the line. Some examples are found in + core/oss/mixer_oss.c, core/oss/and + pcm_oss.c. + + + + For a raw-data proc-file, set the attributes like the following: + + + +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; +]]> + + + + + + The callback is much more complicated than the text-file + version. You need to use a low-level i/o functions such as + copy_from/to_user() to transfer the + data. + + + + local_max_size) + size = local_max_size - pos; + if (copy_to_user(buf, local_data + pos, size)) + return -EFAULT; + return size; + } +]]> + + + + + + + + + + + + Power Management + + If the chip is supposed to work with with suspend/resume + functions, you need to add the power-management codes to the + driver. The additional codes for the power-management should be + ifdef'ed with + CONFIG_PM. + + + + ALSA provides the common power-management layer. Each card driver + needs to have only low-level suspend and resume callbacks. + + + + + + + + + + The scheme of the real suspend job is as following. + + + Retrieve the chip data from pm_private_data field. + Call snd_pcm_suspend_all() to suspend the running PCM streams. + Save the register values if necessary. + Stop the hardware if necessary. + Disable the PCI device by calling pci_disable_device(). + + + + + A typical code would be like: + + + +pm_private_data; + /* (2) */ + snd_pcm_suspend_all(chip->pcm); + /* (3) */ + snd_mychip_save_registers(chip); + /* (4) */ + snd_mychip_stop_hardware(chip); + /* (5) */ + pci_disable_device(chip->pci); + return 0; + } +]]> + + + + + + The scheme of the real resume job is as following. + + + Retrieve the chip data from pm_private_data field. + Enable the pci device again by calling + pci_enable_device(). + Re-initialize the chip. + Restore the saved registers if necessary. + Resume the mixer, e.g. calling + snd_ac97_resume(). + Restart the hardware (if any). + + + + + A typical code would be like: + + + +pm_private_data; + /* (2) */ + pci_enable_device(chip->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); + return 0; + } +]]> + + + + + + OK, we have all callbacks now. Let's set up them now. In the + initialization of the card, add the following: + + + + + + + + Here you don't have to put ifdef CONFIG_PM around, since it's already + checked in the header and expanded to empty if not needed. + + + + If you need a space for saving the registers, you'll need to + 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. + + + + And next, set suspend/resume callbacks to the pci_driver, + This can be done by passing a macro SND_PCI_PM_CALLBACKS + in the pci_driver struct. This macro is expanded to the correct + (global) callbacks if CONFIG_PM is set. + + + + + + + + + + + + + + + + Module Parameters + + There are standard module options for ALSA. At least, each + module should have index, + id and enable + options. + + + + If the module supports multiple cards (usually up to + 8 = SNDRV_CARDS cards), they should be + arrays. The default initial values are defined already as + constants for ease of programming: + + + + + + + + + + If the module supports only a single card, they could be single + variables, instead. enable option is not + always necessary in this case, but it wouldn't be so bad to have a + dummy option for compatibility. + + + + The module parameters must be declared with the standard + module_param()(), + module_param_array()() and + MODULE_PARM_DESC() macros. + + + + The typical coding would be like below: + + + + + + + + + + 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 tainted. + + + + + + + + + + + + + + + + How To Put Your Driver Into ALSA Tree +
+ General + + 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. + + + + Suppose that you'll create a new PCI driver for the card + xyz. The card module name would be + snd-xyz. The new driver is usually put into alsa-driver + tree, alsa-driver/pci 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 alsa-kernel tree (to the corresponding directory, + such as alsa-kernel/pci) and eventually + integrated into Linux 2.6 tree (the directory would be + linux/sound/pci). + + + + In the following sections, the driver code is supposed + to be put into alsa-driver tree. The two cases are assumed: + a driver consisting of a single source file and one consisting + of several source files. + +
+ +
+ Driver with A Single Source File + + + + + Modify alsa-driver/pci/Makefile + + + + Suppose you have a file xyz.c. Add the following + two lines + + + + + + + + + + + Create the Kconfig entry + + + + Add the new entry of Kconfig for your xyz driver. + + + + + + + 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. + + + + 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. + + + + For the details of Kconfig script, refer to the kbuild + documentation. + + + + + + + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + + + + +
+ +
+ Drivers with Several Source Files + + Suppose that the driver snd-xyz have several source files. + They are located in the new subdirectory, + pci/xyz. + + + + + Add a new directory (xyz) in + alsa-driver/pci/Makefile like below + + + + + + + + + + + + Under the directory xyz, create a Makefile + + + Sample Makefile for a driver xyz + + + + + + + + + + Create the Kconfig entry + + + + This procedure is as same as in the last section. + + + + + + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + + + + +
+ + + + + + + + Useful Functions + +
+ <function>snd_printk()</function> and friends + + ALSA provides a verbose version of + printk() function. If a kernel config + CONFIG_SND_VERBOSE_PRINTK is set, this + function prints the given message together with the file name + and the line of the caller. The KERN_XXX + prefix is processed as + well as the original printk() does, so it's + recommended to add this prefix, e.g. + + + + + + + + + + There are also printk()'s for + debugging. snd_printd() can be used for + general debugging purposes. If + CONFIG_SND_DEBUG is set, this function is + compiled, and works just like + snd_printk(). If the ALSA is compiled + without the debugging flag, it's ignored. + + + + snd_printdd() is compiled in only when + CONFIG_SND_DEBUG_DETECT is set. Please note + that DEBUG_DETECT is not set as default + even if you configure the alsa-driver with + option. You need to give + explicitly option instead. + +
+ +
+ <function>snd_assert()</function> + + snd_assert() macro is similar with the + normal assert() macro. For example, + + + + + + + + + + The first argument is the expression to evaluate, and the + second argument is the action if it fails. When + CONFIG_SND_DEBUG, is set, it will show an + error message such as BUG? (xxx) (called from + yyy). When no debug flag is set, this is + ignored. + +
+ +
+ <function>snd_runtime_check()</function> + + This macro is quite similar with + snd_assert(). Unlike + snd_assert(), the expression is always + evaluated regardless of + CONFIG_SND_DEBUG. When + CONFIG_SND_DEBUG is set, the macro will + show a message like ERROR (xx) (called from + yyy). + +
+ +
+ <function>snd_BUG()</function> + + It calls snd_assert(0,) -- that is, just + prints the error message at the point. It's useful to show that + a fatal error happens there. + +
+ + + + + + + + Acknowledgments + + I would like to thank Phil Kerr for his help for improvement and + corrections of this document. + + + Kevin Conder reformatted the original plain-text to the + DocBook format. + + + Giuliano Pochini corrected typos and contributed the example codes + in the hardware constraints section. + + + + + diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt new file mode 100644 index 000000000000..ccda41b10f8a --- /dev/null +++ b/Documentation/sound/alsa/Joystick.txt @@ -0,0 +1,86 @@ +Analog Joystick Support on ALSA Drivers +======================================= + Oct. 14, 2003 + Takashi Iwai + +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 000000000000..5cb970612870 --- /dev/null +++ b/Documentation/sound/alsa/MIXART.txt @@ -0,0 +1,100 @@ + Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards + Digigram + + +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 continously + +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 seperately. + +Capture +------- +There is one substream per capture device. For instance only stereo +formats are supported. + +Mixer +----- + and : analog volume control of playback and capture PCM. + and : digital volume control of each analog substream. + and : digital volume control of each AES/EBU substream. + : 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 +Distributalbe under GPL. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt new file mode 100644 index 000000000000..ec2a02541d5b --- /dev/null +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -0,0 +1,297 @@ + NOTES ON KERNEL OSS-EMULATION + ============================= + + Jan. 22, 2004 Takashi Iwai + + +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. + + +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. + +USB devices +----------- +Some USB devices support only 24bit format packed in 3bytes. This +format is not supported by OSS and no conversion is provided by kernel +OSS emulation. You can use the user-space OSS emulation via libaoss +instead. + diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt new file mode 100644 index 000000000000..25c5d648aef6 --- /dev/null +++ b/Documentation/sound/alsa/Procfile.txt @@ -0,0 +1,191 @@ + Proc Files of ALSA Drivers + ========================== + Takashi Iwai + +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 + -: + +pcm + Lists the currently available PCM devices in format of + -: : : + +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 (desribed 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 + + +Sequencer Information +--------------------- + +seq/drivers + Lists the currently available ALSA sequencer drivers. + +seq/clients + Shows the list of currently available sequencer clinets and + ports. The connection status and the running status are shown + in this file, too. + +seq/queues + Lists the currently allocated/running sequener 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 reigster 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 000000000000..651adaf60473 --- /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 acticle 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 000000000000..1b0ac06ba95d --- /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/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt new file mode 100644 index 000000000000..e9d07b8f1acb --- /dev/null +++ b/Documentation/sound/alsa/hda_codec.txt @@ -0,0 +1,299 @@ +Notes on Universal Interface for Intel High Definition Audio Codec +------------------------------------------------------------------ + +Takashi Iwai + + +[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 *); +}; + +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 last, private_free callback, is optional. It's called in the +destructor to release any necessary data in the lowlevel driver. + +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(snd_card_t *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 list_head *p; + struct hda_codec *codec; + list_for_each(p, &bus->codec_list) { + codec = list_entry(p, struct hda_codec, 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_codec_read() and snd_codec_write(). +snd_hda_param_read() is for reading parameters. +For writing a sequence of verbs, use snd_hda_sequence_write(). + +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, + snd_pcm_substream_t *substream); + int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, + snd_pcm_substream_t *substream); + int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, + unsigned int stream_tag, unsigned int format, + snd_pcm_substream_t *substream); + int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, + snd_pcm_substream_t *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 + }; + +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. + +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. In the patch resume callback, call +snd_hda_resume_spdif(). + + +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 table with struct +hda_board_config 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 snd_kcontrol_new_t. The same array +can be passed to snd_hda_resume_ctls() for resume. +Note that this will call control->put callback of these entries. So, +put callback should check codec->in_resume and force to restore the +given value if it's non-zero even if the value is identical with the +cached value. + +Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be +used for the entry of snd_kcontrol_new_t. + +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/seq_oss.html b/Documentation/sound/alsa/seq_oss.html new file mode 100644 index 000000000000..d9776cf60c07 --- /dev/null +++ b/Documentation/sound/alsa/seq_oss.html @@ -0,0 +1,409 @@ + + + + OSS Sequencer Emulation on ALSA + + + +
+

+ +

+ +
+

+OSS Sequencer Emulation on ALSA

+ +
+

Copyright (c) 1998,1999 by Takashi Iwai +<iwai@ww.uni-erlangen.de> +

ver.0.1.8; Nov. 16, 1999 +

+ +

+ +

+1. Description

+This directory contains the OSS sequencer emulation driver on ALSA. Note +that this program is still in the development state. +

What this does - it provides the emulation of the OSS sequencer, access +via +/dev/sequencer and /dev/music devices. +The most of applications using OSS can run if the appropriate ALSA +sequencer is prepared. +

The following features are emulated by this driver: +

    +
  • +Normal sequencer and MIDI events:
    • + +
    They are converted to the ALSA sequencer events, and sent to the corresponding +port. +
  • +Timer events:
    • + +
    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 /dev/music. + +
  • +Patch loading:
    • + +
    It purely depends on the synth drivers whether it's supported since +the patch loading is realized by callback to the synth driver. +
  • +I/O controls:
    • + +
    Most of controls are accepted. Some controls +are dependent on the synth driver, as well as even on original OSS.
+Furthermore, you can find the following advanced features: +
    +
  • +Better queue mechanism:
    • + +
    The events are queued before processing them. +
  • +Multiple applications:
    • + +
    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. +
  • +Real-time event processing:
    • + +
    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. +
  • +/proc interface:
    • + +
    The status of applications and devices can be shown via /proc/asound/seq/oss +at any time. In the later version, configuration will be changed via /proc +interface, too.
+ +

+2. Installation

+Run configure script with both sequencer support (--with-sequencer=yes) +and OSS emulation (--with-oss=yes) options. A module snd-seq-oss.o +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. +

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. +

The available synth and MIDI devices can be found in proc interface. +Run "cat /proc/asound/seq/oss", and check the devices. For example, +if you use an AWE64 card, you'll see like the following: +

        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
+Note that the device number may be different from the information of +/proc/asound/oss-devices +or ones of the original OSS driver. Use the device number listed in /proc/asound/seq/oss +to play via OSS sequencer emulation. +

+3. Using Synthesizer Devices

+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 /dev/sequencer like sfxload, +too. +

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. +

+4. Using MIDI Devices

+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 /proc/asound/seq/oss. +Be aware that these numbers are mostly different from the list in +/proc/asound/oss-devices. +

+5. Module Options

+The following module options are available: +
    +
  • +maxqlen
    • + +
    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. +
  • +seq_oss_debug
    • + +
    specifies the debug level and accepts zero (= no debug message) or +positive integer. Default value is 0.
+ +

+6. Queue Mechanism

+OSS sequencer emulation uses an ALSA priority queue. The +events from /dev/sequencer are processed and put onto the queue +specified by module option. +

All the events from /dev/sequencer 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. +

The queued events are dispatched to the corresponding ALSA sequencer +ports after scheduled time by ALSA sequencer dispatcher. +

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. +

The input from MIDI devices or echo-back events are stored on read FIFO +queue. If application reads /dev/sequencer in blocking mode, the +process will be awaked. + +

+7. Interface to Synthesizer Device

+ +

+7.1. Registration

+To register an OSS synthesizer device, use snd_seq_oss_synth_register +function. +
int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+                              snd_seq_oss_callback_t *oper, void *private_data)
+The arguments name, type, subtype and +nvoices +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. +

To release this device, call snd_seq_oss_synth_unregister function: +

int snd_seq_oss_synth_unregister(int index),
+where the index is the index number returned by register function. +

+7.2. Callbacks

+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: +
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 open and close callbacks, they are allowed
+to be NULL.
+
Each callback function takes the argument type snd_seq_oss_arg_t as the
+first argument.
+
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;
+};

+The first three fields, app_index, file_mode and
+seq_mode
+are initialized by OSS sequencer. The app_index is the application
+index which is unique to each application opening OSS sequencer. The
+file_mode
+is bit-flags indicating the file operation mode. See
+seq_oss.h
+for its meaning. The seq_mode is sequencer operation mode. In
+the current version, only SND_OSSSEQ_MODE_SYNTH is used.
+
The next two fields, addr and private_data, must be
+filled by the synth driver at open callback. The addr contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for private_data, it must be released
+in close callback by itself.
+
The last field, event_passing, indicates how to translate note-on
+/ off events. In PROCESS_EVENTS mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In PASS_EVENTS
+mode, all note on/off events are passed to the port without modified. PROCESS_KEYPRESS
+mode checks the note above 128 and regards it as key pressure event (mainly
+for Emu8000 driver).
+

+7.2.1. Open Callback

+The open 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:
+
    
+
  1. 
+Allocate private data record.
    2. 
+
+
  2. 
+Create an ALSA sequencer port.
    3. 
+
+
  3. 
+Set the new port address on arg->addr.
    4. 
+
+
  4. 
+Set the private data record pointer on arg->private_data.
    5. 
+

+Note that the type bit-flags in port_info of this synth port must NOT contain
+TYPE_MIDI_GENERIC
+bit. Instead, TYPE_SPECIFIC should be used. Also, CAP_SUBSCRIPTION
+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.
+

+7.2.2 Ioctl Callback

+The ioctl callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+
    
+
  • 
+IOCTL_SEQ_RESET_SAMPLES
    • 
+
+
    reset all samples on memory -- return 0
+
  • 
+IOCTL_SYNTH_MEMAVL
    • 
+
+
    return the available memory size
+
  • 
+FM_4OP_ENABLE
    • 
+
+
    can be ignored usually

+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+

+7.2.3 Load_Patch Callback

+The load_patch 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.
+

+7.2.4 Close Callback

+The close 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.
+

+7.2.5 Reset Callback

+The reset 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 HEARTBEAT event to the
+port.
+

+7.3 Events

+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:
+
 
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
ALSA eventOriginal OSS events
NOTEONSEQ_NOTEON
+
MIDI_NOTEON
NOTESEQ_NOTEOFF
+
MIDI_NOTEOFF
KEYPRESSMIDI_KEY_PRESSURE
CHANPRESSSEQ_AFTERTOUCH
+
MIDI_CHN_PRESSURE
PGMCHANGESEQ_PGMCHANGE
+
MIDI_PGM_CHANGE
PITCHBENDSEQ_CONTROLLER(CTRL_PITCH_BENDER)
+
MIDI_PITCH_BEND
CONTROLLERMIDI_CTL_CHANGE
+
SEQ_BALANCE (with CTL_PAN)
CONTROL14SEQ_CONTROLLER
REGPARAMSEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
SYSEXSEQ_SYSEX

+
+
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.
+
Some OSS events (SEQ_PRIVATE and SEQ_VOLUME 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.
+

+8. Interface to MIDI Device

+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 SND_SEQ_GROUP_DEVICE and a capability-bit CAP_READ or
+CAP_WRITE. Also, subscription capabilities, CAP_SUBS_READ or CAP_SUBS_WRITE,
+must be defined, too. If these conditions are not satisfied, the port is not
+registered as OSS sequencer MIDI device.
+
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.
+

+9. Known Problems / TODO's

+
+
    
+
  • 
+Patch loading via ALSA instrument layer is not implemented yet.
    • 
+

+
+
+
diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt
new file mode 100644
index 000000000000..c1919559d509
--- /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.
-- 
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