diff options
| author | André Fabian Silva Delgado <emulatorman@parabola.nu> | 2015-08-05 17:04:01 -0300 |
|---|---|---|
| committer | André Fabian Silva Delgado <emulatorman@parabola.nu> | 2015-08-05 17:04:01 -0300 |
| commit | 57f0f512b273f60d52568b8c6b77e17f5636edc0 (patch) | |
| tree | 5e910f0e82173f4ef4f51111366a3f1299037a7b /Documentation/sound/alsa | |
Initial import
Diffstat (limited to 'Documentation/sound/alsa')
36 files changed, 9462 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt new file mode 100644 index 000000000..5a8583abe --- /dev/null +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -0,0 +1,2312 @@ + + Advanced Linux Sound Architecture - Driver + ========================================== + Configuration guide + + +Kernel Configuration +==================== + +To enable ALSA support you need at least to build the kernel with +primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS, +you don't have to choose any of the OSS modules. + +Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and +PCM supports if you want to run OSS applications with ALSA. + +If you want to support the WaveTable functionality on cards such as +SB Live! then you need to enable "Sequencer support" +(CONFIG_SND_SEQUENCER). + +To make ALSA debug messages more verbose, enable the "Verbose printk" +and "Debug" options. To check for memory leaks, turn on "Debug memory" +too. "Debug detection" will add checks for the detection of cards. + +Please note that all the ALSA ISA drivers support the Linux isapnp API +(if the card supports ISA PnP). You don't need to configure the cards +using isapnptools. + + +Creating ALSA devices +===================== + +This depends on your distribution, but normally you use the /dev/MAKEDEV +script to create the necessary device nodes. On some systems you use a +script named 'snddevices'. + + +Module parameters +================= + +The user can load modules with options. If the module supports more than +one card and you have more than one card of the same type then you can +specify multiple values for the option separated by commas. + +Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. + + Module snd + ---------- + + The core ALSA module. It is used by all ALSA card drivers. + It takes the following options which have global effects. + + major - major number for sound driver + - Default: 116 + cards_limit + - limiting card index for auto-loading (1-8) + - Default: 1 + - For auto-loading more than one card, specify this + option together with snd-card-X aliases. + slots - Reserve the slot index for the given driver. + This option takes multiple strings. + See "Module Autoloading Support" section for details. + debug - Specifies the debug message level + (0 = disable debug prints, 1 = normal debug messages, + 2 = verbose debug messages) + This option appears only when CONFIG_SND_DEBUG=y. + This option can be dynamically changed via sysfs + /sys/modules/snd/parameters/debug file. + + Module snd-pcm-oss + ------------------ + + The PCM OSS emulation module. + This module takes options which change the mapping of devices. + + dsp_map - PCM device number maps assigned to the 1st OSS device. + - Default: 0 + adsp_map - PCM device number maps assigned to the 2st OSS device. + - Default: 1 + nonblock_open + - Don't block opening busy PCM devices. Default: 1 + + For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of + the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped + to PCM #0 of the card #0. + For changing the second or later card, specify the option with + commas, such like "dsp_map=0,1". + + nonblock_open option is used to change the behavior of the PCM + regarding opening the device. When this option is non-zero, + opening a busy OSS PCM device won't be blocked but return + immediately with EAGAIN (just like O_NONBLOCK flag). + + Module snd-rawmidi + ------------------ + + This module takes options which change the mapping of devices. + similar to those of the snd-pcm-oss module. + + midi_map - MIDI device number maps assigned to the 1st OSS device. + - Default: 0 + amidi_map - MIDI device number maps assigned to the 2st OSS device. + - Default: 1 + + Common parameters for top sound card modules + -------------------------------------------- + + Each of top level sound card module takes the following options. + + index - index (slot #) of sound card + - Values: 0 through 31 or negative + - If nonnegative, assign that index number + - if negative, interpret as a bitmask of permissible + indices; the first free permitted index is assigned + - Default: -1 + id - card ID (identifier or name) + - Can be up to 15 characters long + - Default: the card type + - A directory by this name is created under /proc/asound/ + containing information about the card + - This ID can be used instead of the index number in + identifying the card + enable - enable card + - Default: enabled, for PCI and ISA PnP cards + + Module snd-adlib + ---------------- + + Module for AdLib FM cards. + + port - port # for OPL chip + + This module supports multiple cards. It does not support autoprobe, so + the port must be specified. For actual AdLib FM cards it will be 0x388. + Note that this card does not have PCM support and no mixer; only FM + synthesis. + + Make sure you have "sbiload" from the alsa-tools package available and, + after loading the module, find out the assigned ALSA sequencer port + number through "sbiload -l". Example output: + + Port Client name Port name + 64:0 OPL2 FM synth OPL2 FM Port + + Load the std.sb and drums.sb patches also supplied by sbiload: + + sbiload -p 64:0 std.sb drums.sb + + If you use this driver to drive an OPL3, you can use std.o3 and drums.o3 + instead. To have the card produce sound, use aplaymidi from alsa-utils: + + aplaymidi -p 64:0 foo.mid + + Module snd-ad1816a + ------------------ + + Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. + + clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) + + This module supports multiple cards, autoprobe and PnP. + + Module snd-ad1848 + ----------------- + + Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. + + port - port # for AD1848 chip + irq - IRQ # for AD1848 chip + dma1 - DMA # for AD1848 chip (0,1,3) + + This module supports multiple cards. It does not support autoprobe + thus main port must be specified!!! Other ports are optional. + + The power-management is supported. + + Module snd-ad1889 + ----------------- + + Module for Analog Devices AD1889 chips. + + ac97_quirk - AC'97 workaround for strange hardware + See the description of intel8x0 module for details. + + This module supports multiple cards. + + Module snd-ali5451 + ------------------ + + Module for ALi M5451 PCI chip. + + pcm_channels - Number of hardware channels assigned for PCM + spdif - Support SPDIF I/O + - Default: disabled + + This module supports one chip and autoprobe. + + The power-management is supported. + + Module snd-als100 + ----------------- + + Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. + + This module supports multiple cards, autoprobe and PnP. + + The power-management is supported. + + Module snd-als300 + ----------------- + + Module for Avance Logic ALS300 and ALS300+ + + This module supports multiple cards. + + The power-management is supported. + + Module snd-als4000 + ------------------ + + Module for sound cards based on Avance Logic ALS4000 PCI chip. + + joystick_port - port # for legacy joystick support. + 0 = disabled (default), 1 = auto-detect + + This module supports multiple cards, autoprobe and PnP. + + The power-management is supported. + + Module snd-asihpi + ----------------- + + Module for AudioScience ASI soundcards + + enable_hpi_hwdep - enable HPI hwdep for AudioScience soundcard + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-atiixp + ----------------- + + Module for ATI IXP 150/200/250/400 AC97 controllers. + + ac97_clock - AC'97 clock (default = 48000) + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + ac97_codec - Workaround to specify which AC'97 codec + instead of probing. If this works for you + file a bug with your `lspci -vn` output. + -2 -- Force probing. + -1 -- Default behavior. + 0-2 -- Use the specified codec. + spdif_aclink - S/PDIF transfer over AC-link (default = 1) + + This module supports one card and autoprobe. + + ATI IXP has two different methods to control SPDIF output. One is + over AC-link and another is over the "direct" SPDIF output. The + implementation depends on the motherboard, and you'll need to + choose the correct one via spdif_aclink module option. + + The power-management is supported. + + Module snd-atiixp-modem + ----------------------- + + Module for ATI IXP 150/200/250 AC97 modem controllers. + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-au8810, snd-au8820, snd-au8830 + ----------------------------------------- + + Module for Aureal Vortex, Vortex2 and Advantage device. + + pcifix - Control PCI workarounds + 0 = Disable all workarounds + 1 = Force the PCI latency of the Aureal card to 0xff + 2 = Force the Extend PCI#2 Internal Master for Efficient + Handling of Dummy Requests on the VIA KT133 AGP Bridge + 3 = Force both settings + 255 = Autodetect what is required (default) + + This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware + EQ, mpu401, gameport. A3D and wavetable support are still in development. + Development and reverse engineering work is being coordinated at + http://savannah.nongnu.org/projects/openvortex/ + SPDIF output has a copy of the AC97 codec output, unless you use the + "spdif" pcm device, which allows raw data passthru. + The hardware EQ hardware and SPDIF is only present in the Vortex2 and + Advantage. + + Note: Some ALSA mixer applications don't handle the SPDIF sample rate + control correctly. If you have problems regarding this, try + another ALSA compliant mixer (alsamixer works). + + Module snd-azt1605 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be chosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-azt2316 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be chosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-aw2 + -------------- + + Module for Audiowerk2 sound card + + This module supports multiple cards. + + Module snd-azt2320 + ------------------ + + Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). + + This module supports multiple cards, PnP and autoprobe. + + The power-management is supported. + + Module snd-azt3328 + ------------------ + + Module for sound cards based on Aztech AZF3328 PCI chip. + + joystick - Enable joystick (default off) + + This module supports multiple cards. + + Module snd-bt87x + ---------------- + + Module for video cards based on Bt87x chips. + + digital_rate - Override the default digital rate (Hz) + load_all - Load the driver even if the card model isn't known + + This module supports multiple cards. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + Module snd-ca0106 + ----------------- + + Module for Creative Audigy LS and SB Live 24bit + + This module supports multiple cards. + + + Module snd-cmi8330 + ------------------ + + Module for sound cards based on C-Media CMI8330 ISA chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + wssport - port # for CMI8330 chip (WSS) + wssirq - IRQ # for CMI8330 chip (WSS) + wssdma - first DMA # for CMI8330 chip (WSS) + sbport - port # for CMI8330 chip (SB16) + sbirq - IRQ # for CMI8330 chip (SB16) + sbdma8 - 8bit DMA # for CMI8330 chip (SB16) + sbdma16 - 16bit DMA # for CMI8330 chip (SB16) + fmport - (optional) OPL3 I/O port + mpuport - (optional) MPU401 I/O port + mpuirq - (optional) MPU401 irq # + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-cmipci + ----------------- + + Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. + + mpu_port - port address of MIDI interface (8338 only): + 0x300,0x310,0x320,0x330 = legacy port, + 0 = disable (default) + fm_port - port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), + 0 = disable + soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) + (default = 1) + joystick_port - Joystick port address (0 = disable, 1 = auto-detect) + + This module supports autoprobe and multiple cards. + + The power-management is supported. + + Module snd-cs4231 + ----------------- + + Module for sound cards based on CS4231 ISA chips. + + port - port # for CS4231 chip + mpu_port - port # for MPU-401 UART (optional), -1 = disable + irq - IRQ # for CS4231 chip + mpu_irq - IRQ # for MPU-401 UART + dma1 - first DMA # for CS4231 chip + dma2 - second DMA # for CS4231 chip + + This module supports multiple cards. This module does not support autoprobe + thus main port must be specified!!! Other ports are optional. + + The power-management is supported. + + Module snd-cs4236 + ----------------- + + Module for sound cards based on CS4232/CS4232A, + CS4235/CS4236/CS4236B/CS4237B/ + CS4238B/CS4239 ISA chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for CS4236 chip (PnP setup - 0x534) + cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) + mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable + fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable + irq - IRQ # for CS4236 chip (5,7,9,11,12,15) + mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) + dma1 - first DMA # for CS4236 chip (0,1,3) + dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable + + This module supports multiple cards. This module does not support autoprobe + (if ISA PnP is not used) thus main port and control port must be + specified!!! Other ports are optional. + + The power-management is supported. + + This module is aliased as snd-cs4232 since it provides the old + snd-cs4232 functionality, too. + + Module snd-cs4281 + ----------------- + + Module for Cirrus Logic CS4281 soundchip. + + dual_codec - Secondary codec ID (0 = disable, default) + + This module supports multiple cards. + + The power-management is supported. + + Module snd-cs46xx + ----------------- + + Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ + CS4624/CS4630/CS4280 PCI chips. + + external_amp - Force to enable external amplifier. + thinkpad - Force to enable Thinkpad's CLKRUN control. + mmap_valid - Support OSS mmap mode (default = 0). + + This module supports multiple cards and autoprobe. + Usually external amp and CLKRUN controls are detected automatically + from PCI sub vendor/device ids. If they don't work, give the options + above explicitly. + + The power-management is supported. + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + + Module snd-cs5535audio + ---------------------- + + Module for multifunction CS5535 companion PCI device + + The power-management is supported. + + Module snd-ctxfi + ---------------- + + Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) + * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series + * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series + * Creative Sound Blaster X-Fi Titanium Professional Audio + * Creative Sound Blaster X-Fi Titanium + * Creative Sound Blaster X-Fi Elite Pro + * Creative Sound Blaster X-Fi Platinum + * Creative Sound Blaster X-Fi Fatal1ty + * Creative Sound Blaster X-Fi XtremeGamer + * Creative Sound Blaster X-Fi XtremeMusic + + reference_rate - reference sample rate, 44100 or 48000 (default) + multiple - multiple to ref. sample rate, 1 or 2 (default) + subsystem - override the PCI SSID for probing; the value + consists of SSVID << 16 | SSDID. The default is + zero, which means no override. + + This module supports multiple cards. + + Module snd-darla20 + ------------------ + + Module for Echoaudio Darla20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-darla24 + ------------------ + + Module for Echoaudio Darla24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-dt019x + ----------------- + + Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP + only) + + This module supports multiple cards. This module is enabled only with + ISA PnP support. + + The power-management is supported. + + Module snd-dummy + ---------------- + + Module for the dummy sound card. This "card" doesn't do any output + or input, but you may use this module for any application which + requires a sound card (like RealPlayer). + + pcm_devs - Number of PCM devices assigned to each card + (default = 1, up to 4) + pcm_substreams - Number of PCM substreams assigned to each PCM + (default = 8, up to 128) + hrtimer - Use hrtimer (=1, default) or system timer (=0) + fake_buffer - Fake buffer allocations (default = 1) + + When multiple PCM devices are created, snd-dummy gives different + behavior to each PCM device: + 0 = interleaved with mmap support + 1 = non-interleaved with mmap support + 2 = interleaved without mmap + 3 = non-interleaved without mmap + + As default, snd-dummy drivers doesn't allocate the real buffers + but either ignores read/write or mmap a single dummy page to all + buffer pages, in order to save the resources. If your apps need + the read/ written buffer data to be consistent, pass fake_buffer=0 + option. + + The power-management is supported. + + Module snd-echo3g + ----------------- + + Module for Echoaudio 3G cards (Gina3G/Layla3G) + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-emu10k1 + ------------------ + + Module for EMU10K1/EMU10k2 based PCI sound cards. + * Sound Blaster Live! + * Sound Blaster PCI 512 + * Emu APS (partially supported) + * Sound Blaster Audigy + + extin - bitmap of available external inputs for FX8010 (see bellow) + extout - bitmap of available external outputs for FX8010 (see bellow) + seq_ports - allocated sequencer ports (4 by default) + max_synth_voices - limit of voices used for wavetable (64 by default) + max_buffer_size - specifies the maximum size of wavetable/pcm buffers + given in MB unit. Default value is 128. + enable_ir - enable IR + + This module supports multiple cards and autoprobe. + + Input & Output configurations [extin/extout] + * Creative Card wo/Digital out [0x0003/0x1f03] + * Creative Card w/Digital out [0x0003/0x1f0f] + * Creative Card w/Digital CD in [0x000f/0x1f0f] + * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] + * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] + * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] + * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] + * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] + * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] + * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] + * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] + * Creative Card all ins and outs [0x3fff/0x7fff] + + The power-management is supported. + + Module snd-emu10k1x + ------------------- + + Module for Creative Emu10k1X (SB Live Dell OEM version) + + This module supports multiple cards. + + Module snd-ens1370 + ------------------ + + Module for Ensoniq AudioPCI ES1370 PCI sound cards. + * SoundBlaster PCI 64 + * SoundBlaster PCI 128 + + joystick - Enable joystick (default off) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-ens1371 + ------------------ + + Module for Ensoniq AudioPCI ES1371 PCI sound cards. + * SoundBlaster PCI 64 + * SoundBlaster PCI 128 + * SoundBlaster Vibra PCI + + joystick_port - port # for joystick (0x200,0x208,0x210,0x218), + 0 = disable (default), 1 = auto-detect + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-es1688 + ----------------- + + Module for ESS AudioDrive ES-1688 and ES-688 sound cards. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) + mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + fm_port - port # for OPL3 (option; share the same port as default) + + with isapnp=0, the following additional options are available: + port - port # for ES-1688 chip (0x220,0x240,0x260) + irq - IRQ # for ES-1688 chip (5,7,9,10) + dma8 - DMA # for ES-1688 chip (0,1,3) + + This module supports multiple cards and autoprobe (without MPU-401 port) + and PnP with the ES968 chip. + + Module snd-es18xx + ----------------- + + Module for ESS AudioDrive ES-18xx sound cards. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for ES-18xx chip (0x220,0x240,0x260) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) + fm_port - port # for FM (optional, not used) + irq - IRQ # for ES-18xx chip (5,7,9,10) + dma1 - first DMA # for ES-18xx chip (0,1,3) + dma2 - first DMA # for ES-18xx chip (0,1,3) + + This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 + port if native ISA PnP routines are not used). + When dma2 is equal with dma1, the driver works as half-duplex. + + The power-management is supported. + + Module snd-es1938 + ----------------- + + Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-es1968 + ----------------- + + Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. + + total_bufsize - total buffer size in kB (1-4096kB) + pcm_substreams_p - playback channels (1-8, default=2) + pcm_substreams_c - capture channels (1-8, default=0) + clock - clock (0 = auto-detection) + use_pm - support the power-management (0 = off, 1 = on, + 2 = auto (default)) + enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default)) + joystick - enable joystick (default off) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-fm801 + ---------------- + + Module for ForteMedia FM801 based PCI sound cards. + + tea575x_tuner - Enable TEA575x tuner + - 1 = MediaForte 256-PCS + - 2 = MediaForte 256-PCPR + - 3 = MediaForte 64-PCR + - High 16-bits are video (radio) device number + 1 + - example: 0x10002 (MediaForte 256-PCPR, device 1) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-gina20 + ----------------- + + Module for Echoaudio Gina20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-gina24 + ----------------- + + Module for Echoaudio Gina24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-gusclassic + --------------------- + + Module for Gravis UltraSound Classic sound card. + + port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for GF1 chip (3,5,9,11,12,15) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe. + + Module snd-gusextreme + --------------------- + + Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. + + port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) + gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) + mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable + irq - IRQ # for ES-1688 chip (5,7,9,10) + gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15) + mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + dma8 - DMA # for ES-1688 chip (0,1,3) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe (without MPU-401 port). + + Module snd-gusmax + ----------------- + + Module for Gravis UltraSound MAX sound card. + + port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for GF1 chip (3,5,9,11,12,15) + dma1 - DMA # for GF1 chip (1,3,5,6,7) + dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + voices - GF1 voices limit (14-32) + pcm_voices - reserved PCM voices + + This module supports multiple cards and autoprobe. + + Module snd-hda-intel + -------------------- + + Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, + PCH, SCH), + ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, + RV630, RV635, RV670, RV770, + VIA VT8251/VT8237A, + SIS966, ULI M5461 + + [Multiple options for each card instance] + model - force the model name + position_fix - Fix DMA pointer + -1 = system default: choose appropriate one per controller + hardware + 0 = auto: falls back to LPIB when POSBUF doesn't work + 1 = use LPIB + 2 = POSBUF: use position buffer + 3 = VIACOMBO: VIA-specific workaround for capture + 4 = COMBO: use LPIB for playback, auto for capture stream + probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + When the bit 8 (0x100) is set, the lower 8 bits are used + as the "fixed" codec slots; i.e. the driver probes the + slots regardless what hardware reports back + probe_only - Only probing and no codec initialization (default=off); + Useful to check the initial codec status for debugging + bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. + Passing -1 will make the driver to choose the appropriate + value based on the controller chip. + patch - Specifies the early "patch" files to modify the HD-audio + setup before initializing the codecs. This option is + available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. + See HD-Audio.txt for details. + beep_mode - Selects the beep registration mode (0=off, 1=on); default + value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig. + + [Single (global) options] + single_cmd - Use single immediate commands to communicate with + codecs (for debugging only) + enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) + power_save - Automatic power-saving timeout (in second, 0 = + disable) + power_save_controller - Reset HD-audio controller in power-saving mode + (default = on) + align_buffer_size - Force rounding of buffer/period sizes to multiples + of 128 bytes. This is more efficient in terms of memory + access but isn't required by the HDA spec and prevents + users from specifying exact period/buffer sizes. + (default = on) + snoop - Enable/disable snooping (default = on) + + This module supports multiple cards and autoprobe. + + See Documentation/sound/alsa/HD-Audio.txt for more details about + HD-audio driver. + + Each codec may have a model table for different configurations. + If your machine isn't listed there, the default (usually minimal) + configuration is set up. You can pass "model=<name>" option to + specify a certain model in such a case. There are different + models depending on the codec chip. The list of available models + is found in HD-Audio-Models.txt + + The model name "generic" is treated as a special case. When this + model is given, the driver uses the generic codec parser without + "codec-patch". It's sometimes good for testing and debugging. + + If the default configuration doesn't work and one of the above + matches with your device, report it together with alsa-info.sh + output (with --no-upload option) to kernel bugzilla or alsa-devel + ML (see the section "Links and Addresses"). + + power_save and power_save_controller options are for power-saving + mode. See powersave.txt for details. + + Note 2: If you get click noises on output, try the module option + position_fix=1 or 2. position_fix=1 will use the SD_LPIB + register value without FIFO size correction as the current + DMA pointer. position_fix=2 will make the driver to use + the position buffer instead of reading SD_LPIB register. + (Usually SD_LPIB register is more accurate than the + position buffer.) + + position_fix=3 is specific to VIA devices. The position + of the capture stream is checked from both LPIB and POSBUF + values. position_fix=4 is a combination mode, using LPIB + for playback and POSBUF for capture. + + NB: If you get many "azx_get_response timeout" messages at + loading, it's likely a problem of interrupts (e.g. ACPI irq + routing). Try to boot with options like "pci=noacpi". Also, you + can try "single_cmd=1" module option. This will switch the + communication method between HDA controller and codecs to the + single immediate commands instead of CORB/RIRB. Basically, the + single command mode is provided only for BIOS, and you won't get + unsolicited events, too. But, at least, this works independently + from the irq. Remember this is a last resort, and should be + avoided as much as possible... + + MORE NOTES ON "azx_get_response timeout" PROBLEMS: + On some hardware, you may need to add a proper probe_mask option + to avoid the "azx_get_response timeout" problem above, instead. + This occurs when the access to non-existing or non-working codec slot + (likely a modem one) causes a stall of the communication via HD-audio + bus. You can see which codec slots are probed by enabling + CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec + proc files. Then limit the slots to probe by probe_mask option. + For example, probe_mask=1 means to probe only the first slot, and + probe_mask=4 means only the third slot. + + The power-management is supported. + + Module snd-hdsp + --------------- + + Module for RME Hammerfall DSP audio interface(s) + + This module supports multiple cards. + + Note: The firmware data can be automatically loaded via hotplug + when CONFIG_FW_LOADER is set. Otherwise, you need to load + the firmware via hdsploader utility included in alsa-tools + package. + The firmware data is found in alsa-firmware package. + + Note: snd-page-alloc module does the job which snd-hammerfall-mem + module did formerly. It will allocate the buffers in advance + when any HDSP cards are found. To make the buffer + allocation sure, load snd-page-alloc module in the early + stage of boot sequence. See "Early Buffer Allocation" + section. + + Module snd-hdspm + ---------------- + + Module for RME HDSP MADI board. + + precise_ptr - Enable precise pointer, or disable. + line_outs_monitor - Send playback streams to analog outs by default. + enable_monitor - Enable Analog Out on Channel 63/64 by default. + + See hdspm.txt for details. + + Module snd-ice1712 + ------------------ + + Module for Envy24 (ICE1712) based PCI sound cards. + * MidiMan M Audio Delta 1010 + * MidiMan M Audio Delta 1010LT + * MidiMan M Audio Delta DiO 2496 + * MidiMan M Audio Delta 66 + * MidiMan M Audio Delta 44 + * MidiMan M Audio Delta 410 + * MidiMan M Audio Audiophile 2496 + * TerraTec EWS 88MT + * TerraTec EWS 88D + * TerraTec EWX 24/96 + * TerraTec DMX 6Fire + * TerraTec Phase 88 + * Hoontech SoundTrack DSP 24 + * Hoontech SoundTrack DSP 24 Value + * Hoontech SoundTrack DSP 24 Media 7.1 + * Event Electronics, EZ8 + * Digigram VX442 + * Lionstracs, Mediastaton + * Terrasoniq TS 88 + + model - Use the given board model, one of the following: + delta1010, dio2496, delta66, delta44, audiophile, delta410, + delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, + dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, + phase88, mediastation + omni - Omni I/O support for MidiMan M-Audio Delta44/66 + cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver) + in msec resolution, default value is 500 (0.5 sec) + + This module supports multiple cards and autoprobe. Note: The consumer part + is not used with all Envy24 based cards (for example in the MidiMan Delta + serie). + + Note: The supported board is detected by reading EEPROM or PCI + SSID (if EEPROM isn't available). You can override the + model by passing "model" module option in case that the + driver isn't configured properly or you want to try another + type for testing. + + Module snd-ice1724 + ------------------ + + Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. + * MidiMan M Audio Revolution 5.1 + * MidiMan M Audio Revolution 7.1 + * MidiMan M Audio Audiophile 192 + * AMP Ltd AUDIO2000 + * TerraTec Aureon 5.1 Sky + * TerraTec Aureon 7.1 Space + * TerraTec Aureon 7.1 Universe + * TerraTec Phase 22 + * TerraTec Phase 28 + * AudioTrak Prodigy 7.1 + * AudioTrak Prodigy 7.1 LT + * AudioTrak Prodigy 7.1 XT + * AudioTrak Prodigy 7.1 HIFI + * AudioTrak Prodigy 7.1 HD2 + * AudioTrak Prodigy 192 + * Pontis MS300 + * Albatron K8X800 Pro II + * Chaintech ZNF3-150 + * Chaintech ZNF3-250 + * Chaintech 9CJS + * Chaintech AV-710 + * Shuttle SN25P + * Onkyo SE-90PCI + * Onkyo SE-200PCI + * ESI Juli@ + * ESI Maya44 + * Hercules Fortissimo IV + * EGO-SYS WaveTerminal 192M + + model - Use the given board model, one of the following: + revo51, revo71, amp2000, prodigy71, prodigy71lt, + prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, + juli, aureon51, aureon71, universe, ap192, k8x800, + phase22, phase28, ms300, av710, se200pci, se90pci, + fortissimo4, sn25p, WT192M, maya44 + + This module supports multiple cards and autoprobe. + + Note: The supported board is detected by reading EEPROM or PCI + SSID (if EEPROM isn't available). You can override the + model by passing "model" module option in case that the + driver isn't configured properly or you want to try another + type for testing. + + Module snd-indigo + ----------------- + + Module for Echoaudio Indigo + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-indigodj + ------------------- + + Module for Echoaudio Indigo DJ + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-indigoio + ------------------- + + Module for Echoaudio Indigo IO + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-intel8x0 + ------------------- + + Module for AC'97 motherboards from Intel and compatibles. + * Intel i810/810E, i815, i820, i830, i84x, MX440 + ICH5, ICH6, ICH7, 6300ESB, ESB2 + * SiS 7012 (SiS 735) + * NVidia NForce, NForce2, NForce3, MCP04, CK804 + CK8, CK8S, MCP501 + * AMD AMD768, AMD8111 + * ALi m5455 + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + buggy_irq - Enable workaround for buggy interrupts on some + motherboards (default yes on nForce chips, + otherwise off) + buggy_semaphore - Enable workaround for hardware with buggy + semaphores (e.g. on some ASUS laptops) + (default off) + spdif_aclink - Use S/PDIF over AC-link instead of direct connection + from the controller chip + (0 = off, 1 = on, -1 = default) + + This module supports one chip and autoprobe. + + Note: the latest driver supports auto-detection of chip clock. + if you still encounter too fast playback, specify the clock + explicitly via the module option "ac97_clock=41194". + + Joystick/MIDI ports are not supported by this driver. If your + motherboard has these devices, use the ns558 or snd-mpu401 + modules, respectively. + + The power-management is supported. + + Module snd-intel8x0m + -------------------- + + Module for Intel ICH (i8x0) chipset MC97 modems. + * Intel i810/810E, i815, i820, i830, i84x, MX440 + ICH5, ICH6, ICH7 + * SiS 7013 (SiS 735) + * NVidia NForce, NForce2, NForce2s, NForce3 + * AMD AMD8111 + * ALi m5455 + + ac97_clock - AC'97 codec clock base (0 = auto-detect) + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-interwave + -------------------- + + Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 + and other sound cards based on AMD InterWave (tm) chip. + + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) + pcm_voices - reserved PCM voices for the synthesizer (default 2) + effect - 1 = InterWave effects enable (default 0); + requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + + This module supports multiple cards, autoprobe and ISA PnP. + + Module snd-interwave-stb + ------------------------ + + Module for UltraSound 32-Pro (sound card from STB used by Compaq) + and other sound cards based on AMD InterWave (tm) chip with TEA6330T + circuit for extended control of bass, treble and master volume. + + joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) + midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) + pcm_voices - reserved PCM voices for the synthesizer (default 2) + effect - 1 = InterWave effects enable (default 0); + requires 8 voices + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) + port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) + irq - IRQ # for InterWave chip (3,5,9,11,12,15) + dma1 - DMA # for InterWave chip (0,1,3,5,6,7) + dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + + This module supports multiple cards, autoprobe and ISA PnP. + + Module snd-jazz16 + ------------------- + + Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: + MVD1216 + MVA416 + MVA514. + + port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for SB DSP chip (3,5,7,9,10,15) + dma8 - DMA # for SB DSP chip (1,3) + dma16 - DMA # for SB DSP chip (5,7) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (2,3,5,7) + + This module supports multiple cards. + + Module snd-korg1212 + ------------------- + + Module for Korg 1212 IO PCI card + + This module supports multiple cards. + + Module snd-layla20 + ------------------ + + Module for Echoaudio Layla20 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-layla24 + ------------------ + + Module for Echoaudio Layla24 + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-lola + --------------- + + Module for Digigram Lola PCI-e boards + + This module supports multiple cards. + + Module snd-lx6464es + ------------------- + + Module for Digigram LX6464ES boards + + This module supports multiple 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) + + This module supports autoprobe and multiple chips. + + Note: the binding of amplifier is dependent on hardware. + If there is no sound even though all channels are unmuted, try to + specify other gpio connection via amp_gpio option. + For example, a Panasonic notebook might need "amp_gpio=0x0d" + option. + + The power-management is supported. + + Module snd-mia + --------------- + + Module for Echoaudio Mia + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-miro + --------------- + + Module for Miro soundcards: miroSOUND PCM 1 pro, + miroSOUND PCM 12, + miroSOUND PCM 20 Radio. + + port - Port # (0x530,0x604,0xe80,0xf40) + irq - IRQ # (5,7,9,10,11) + dma1 - 1st dma # (0,1,3) + dma2 - 2nd dma # (0,1) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (5,7,9,10) + fm_port - FM Port # (0x388) + wss - enable WSS mode + ide - enable onboard ide support + + Module snd-mixart + ----------------- + + Module for Digigram miXart8 sound cards. + + This module supports multiple cards. + Note: One miXart8 board will be represented as 4 alsa cards. + See MIXART.txt for details. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via mixartloader utility in alsa-tools package. + + Module snd-mona + --------------- + + Module for Echoaudio Mona + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + + Module snd-mpu401 + ----------------- + + Module for MPU-401 UART devices. + + port - port number or -1 (disable) + irq - IRQ number or -1 (disable) + pnp - PnP detection - 0 = disable, 1 = enable (default) + + This module supports multiple devices and PnP. + + Module snd-msnd-classic + ----------------------- + + Module for Turtle Beach MultiSound Classic, Tahiti or Monterey + soundcards. + + io - Port # for msnd-classic card + irq - IRQ # for msnd-classic card + mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, + 0xe0000 or 0xe8000) + write_ndelay - enable write ndelay (default = 1) + calibrate_signal - calibrate signal (default = 0) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + digital - Digital daughterboard present (default = 0) + cfg - Config port (0x250, 0x260 or 0x270) default = PnP + reset - Reset all devices + mpu_io - MPU401 I/O port + mpu_irq - MPU401 irq# + ide_io0 - IDE port #0 + ide_io1 - IDE port #1 + ide_irq - IDE irq# + joystick_io - Joystick I/O port + + The driver requires firmware files "/*(DEBLOBBED)*/" and + "/*(DEBLOBBED)*/" in the proper firmware directory. + + See Documentation/sound/oss/MultiSound for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at + http://www.turtlebeach.com + + Module snd-msnd-pinnacle + ------------------------ + + Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. + + io - Port # for pinnacle/fiji card + irq - IRQ # for pinnalce/fiji card + mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, + 0xe0000 or 0xe8000) + write_ndelay - enable write ndelay (default = 1) + calibrate_signal - calibrate signal (default = 0) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + The driver requires firmware files "/*(DEBLOBBED)*/" and + "/*(DEBLOBBED)*/" in the proper firmware directory. + + Module snd-mtpav + ---------------- + + Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel + port). + + port - I/O port # for MTPAV (0x378,0x278, default=0x378) + irq - IRQ # for MTPAV (7,5, default=7) + hwports - number of supported hardware ports, default=8. + + Module supports only 1 card. This module has no enable option. + + Module snd-mts64 + ---------------- + + Module for Ego Systems (ESI) Miditerminal 4140 + + This module supports multiple devices. + Requires parport (CONFIG_PARPORT). + + Module snd-nm256 + ---------------- + + Module for NeoMagic NM256AV/ZX chips + + playback_bufsize - max playback frame size in kB (4-128kB) + capture_bufsize - max capture frame size in kB (4-128kB) + force_ac97 - 0 or 1 (disabled by default) + buffer_top - specify buffer top address + use_cache - 0 or 1 (disabled by default) + vaio_hack - alias buffer_top=0x25a800 + reset_workaround - enable AC97 RESET workaround for some laptops + reset_workaround2 - enable extended AC97 RESET workaround for some + other laptops + + This module supports one chip and autoprobe. + + The power-management is supported. + + Note: on some notebooks the buffer address cannot be detected + automatically, or causes hang-up during initialization. + In such a case, specify the buffer top address explicitly via + the buffer_top option. + For example, + Sony F250: buffer_top=0x25a800 + Sony F270: buffer_top=0x272800 + The driver supports only ac97 codec. It's possible to force + to initialize/use ac97 although it's not detected. In such a + case, use force_ac97=1 option - but *NO* guarantee whether it + works! + + Note: The NM256 chip can be linked internally with non-AC97 + codecs. This driver supports only the AC97 codec, and won't work + with machines with other (most likely CS423x or OPL3SAx) chips, + even though the device is detected in lspci. In such a case, try + other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP + but some doesn't have ISA PnP. You'll need to specify isapnp=0 + and proper hardware parameters in the case without ISA PnP. + + Note: some laptops need a workaround for AC97 RESET. For the + known hardware like Dell Latitude LS and Sony PCG-F305, this + workaround is enabled automatically. For other laptops with a + hard freeze, you can try reset_workaround=1 option. + + Note: Dell Latitude CSx laptops have another problem regarding + AC97 RESET. On these laptops, reset_workaround2 option is + turned on as default. This option is worth to try if the + previous reset_workaround option doesn't help. + + Note: This driver is really crappy. It's a porting from the + OSS driver, which is a result of black-magic reverse engineering. + The detection of codec will fail if the driver is loaded *after* + X-server as described above. You might be able to force to load + the module, but it may result in hang-up. Hence, make sure that + you load this module *before* X if you encounter this kind of + problem. + + Module snd-opl3sa2 + ------------------ + + Module for Yamaha OPL3-SA2/SA3 sound cards. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - control port # for OPL3-SA chip (0x370) + sb_port - SB port # for OPL3-SA chip (0x220,0x240) + wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) + midi_port - port # for MPU-401 UART (0x300,0x330), -1 = disable + fm_port - FM port # for OPL3-SA chip (0x388), -1 = disable + irq - IRQ # for OPL3-SA chip (5,7,9,10) + dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3) + dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable + + This module supports multiple cards and ISA PnP. It does not support + autoprobe (if ISA PnP is not used) thus all ports must be specified!!! + + The power-management is supported. + + Module snd-opti92x-ad1848 + ------------------------- + + Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. + Module works with OAK Mozart cards as well. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-opti92x-cs4231 + ------------------------- + + Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + dma2 - second DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-opti93x + ------------------ + + Module for sound cards based on OPTi 82c93x chips. + + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for WSS chip (0x530,0xe80,0xf40,0x604) + mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) + fm_port - port # for OPL3 device (0x388) + irq - IRQ # for WSS chip (5,7,9,10,11) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) + dma1 - first DMA # for WSS chip (0,1,3) + dma2 - second DMA # for WSS chip (0,1,3) + + This module supports only one card, autoprobe and PnP. + + Module snd-oxygen + ----------------- + + Module for sound cards based on the C-Media CMI8786/8787/8788 chip: + * Asound A-8788 + * Asus Xonar DG/DGX + * AuzenTech X-Meridian + * AuzenTech X-Meridian 2G + * Bgears b-Enspirer + * Club3D Theatron DTS + * HT-Omega Claro (plus) + * HT-Omega Claro halo (XT) + * Kuroutoshikou CMI8787-HG2PCI + * Razer Barracuda AC-1 + * Sondigo Inferno + * TempoTec HiFier Fantasia + * TempoTec HiFier Serenade + + This module supports autoprobe and multiple cards. + + Module snd-pcsp + ----------------- + + Module for internal PC-Speaker. + + nopcm - Disable PC-Speaker PCM sound. Only beeps remain. + nforce_wa - enable NForce chipset workaround. Expect bad sound. + + This module supports system beeps, some kind of PCM playback and + even a few mixer controls. + + Module snd-pcxhr + ---------------- + + Module for Digigram PCXHR boards + + This module supports multiple cards. + + Module snd-portman2x4 + --------------------- + + Module for Midiman Portman 2x4 parallel port MIDI interface + + This module supports multiple cards. + + Module snd-powermac (on ppc only) + --------------------------------- + + Module for PowerMac, iMac and iBook on-board soundchips + + enable_beep - enable beep using PCM (enabled as default) + + Module supports autoprobe a chip. + + Note: the driver may have problems regarding endianness. + + The power-management is supported. + + Module snd-pxa2xx-ac97 (on arm only) + ------------------------------------ + + Module for AC97 driver for the Intel PXA2xx chip + + For ARM architecture only. + + The power-management is supported. + + Module snd-riptide + ------------------ + + Module for Conexant Riptide chip + + joystick_port - Joystick port # (default: 0x200) + mpu_port - MPU401 port # (default: 0x330) + opl3_port - OPL3 port # (default: 0x388) + + This module supports multiple cards. + The driver requires the firmware loader support on kernel. + You need to install the firmware file "/*(DEBLOBBED)*/" to the standard + firmware path (e.g. /lib/firmware). + + Module snd-rme32 + ---------------- + + Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, + Prodif96 and Prodif Gold) sound cards. + + This module supports multiple cards. + + Module snd-rme96 + ---------------- + + Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. + + This module supports multiple cards. + + Module snd-rme9652 + ------------------ + + Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. + + precise_ptr - Enable precise pointer (doesn't work reliably). + (default = 0) + + This module supports multiple cards. + + Note: snd-page-alloc module does the job which snd-hammerfall-mem + module did formerly. It will allocate the buffers in advance + when any RME9652 cards are found. To make the buffer + allocation sure, load snd-page-alloc module in the early + stage of boot sequence. See "Early Buffer Allocation" + section. + + Module snd-sa11xx-uda1341 (on arm only) + --------------------------------------- + + Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. + + Module supports only one card. + Module has no enable and index options. + + The power-management is supported. + + Module snd-sb8 + -------------- + + Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, + SoundBlaster 2.0, + SoundBlaster Pro + + port - port # for SB DSP chip (0x220,0x240,0x260) + irq - IRQ # for SB DSP chip (5,7,9,10) + dma8 - DMA # for SB DSP chip (1,3) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-sb16 and snd-sbawe + ----------------------------- + + Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), + SoundBlaster AWE 32 (PnP), + SoundBlaster AWE 64 PnP + + mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) + csp - ASP/CSP chip support - 0 = disable (default), 1 = enable + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + port - port # for SB DSP 4.x chip (0x220,0x240,0x260) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable + awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) + (snd-sbawe module only) + irq - IRQ # for SB DSP 4.x chip (5,7,9,10) + dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) + dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) + + This module supports multiple cards, autoprobe and ISA PnP. + + Note: To use Vibra16X cards in 16-bit half duplex mode, you must + disable 16bit DMA with dma16 = -1 module parameter. + Also, all Sound Blaster 16 type cards can operate in 16-bit + half duplex mode through 8-bit DMA channel by disabling their + 16-bit DMA channel. + + The power-management is supported. + + Module snd-sc6000 + ----------------- + + Module for Gallant SC-6000 soundcard and later models: SC-6600 + and SC-7000. + + port - Port # (0x220 or 0x240) + mss_port - MSS Port # (0x530 or 0xe80) + irq - IRQ # (5,7,9,10,11) + mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq + dma - DMA # (1,3,0) + joystick - Enable gameport - 0 = disable (default), 1 = enable + + This module supports multiple cards. + + This card is also known as Audio Excel DSP 16 or Zoltrix AV302. + + Module snd-sscape + ----------------- + + Module for ENSONIQ SoundScape cards. + + port - Port # (PnP setup) + wss_port - WSS Port # (PnP setup) + irq - IRQ # (PnP setup) + mpu_irq - MPU-401 IRQ # (PnP setup) + dma - DMA # (PnP setup) + dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable + + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. + + Module snd-sun-amd7930 (on sparc only) + -------------------------------------- + + Module for AMD7930 sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-sun-cs4231 (on sparc only) + ------------------------------------- + + Module for CS4231 sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-sun-dbri (on sparc only) + ----------------------------------- + + Module for DBRI sound chips found on Sparcs. + + This module supports multiple cards. + + Module snd-wavefront + -------------------- + + Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + + use_cs4232_midi - Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + with isapnp=0, the following options are available: + + cs4232_pcm_port - Port # for CS4232 PCM interface. + cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). + cs4232_mpu_port - Port # for CS4232 MPU-401 interface. + cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). + ics2115_port - Port # for ICS2115 + ics2115_irq - IRQ # for ICS2115 + fm_port - FM OPL-3 Port # + dma1 - DMA1 # for CS4232 PCM interface. + dma2 - DMA2 # for CS4232 PCM interface. + + The below are options for wavefront_synth features: + wf_raw - Assume that we need to boot the OS (default:no) + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. + fx_raw - Assume that the FX process needs help (default:yes) + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. + debug_default - Debug parameters for card initialization + wait_usecs - How long to wait without sleeping, usecs + (default:150) + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. + sleep_interval - How long to sleep when waiting for reply + (default: 100) + sleep_tries - How many times to try sleeping during a wait + (default: 50) + /*(DEBLOBBED)*/ + reset_time - How long to wait for a reset to take effect + (default:2) + ramcheck_time - How many seconds to wait for the RAM test + (default:20) + osrun_time - How many seconds to wait for the ICS2115 OS + (default:10) + + This module supports multiple cards and ISA PnP. + + /*(DEBLOBBED)*/. + + Module snd-sonicvibes + --------------------- + + Module for S3 SonicVibes PCI sound cards. + * PINE Schubert 32 PCI + + reverb - Reverb Enable - 1 = enable, 0 = disable (default) + - SoundCard must have onboard SRAM for this. + mge - Mic Gain Enable - 1 = enable, 0 = disable (default) + + This module supports multiple cards and autoprobe. + + Module snd-serial-u16550 + ------------------------ + + Module for UART16550A serial MIDI ports. + + port - port # for UART16550A chip + irq - IRQ # for UART16550A chip, -1 = poll mode + speed - speed in bauds (9600,19200,38400,57600,115200) + 38400 = default + base - base for divisor in bauds (57600,115200,230400,460800) + 115200 = default + outs - number of MIDI ports in a serial port (1-4) + 1 = default + adaptor - Type of adaptor. + 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, + 3 = MS-124W M/B, 4 = Generic + + This module supports multiple cards. This module does not support autoprobe + thus the main port must be specified!!! Other options are optional. + + Module snd-trident + ------------------ + + Module for Trident 4DWave DX/NX sound cards. + * Best Union Miss Melody 4DWave PCI + * HIS 4DWave PCI + * Warpspeed ONSpeed 4DWave PCI + * AzTech PCI 64-Q3D + * Addonics SV 750 + * CHIC True Sound 4Dwave + * Shark Predator4D-PCI + * Jaton SonicWave 4D + * SiS SI7018 PCI Audio + * Hoontech SoundTrack Digital 4DWave NX + + pcm_channels - max channels (voices) reserved for PCM + wavetable_size - max wavetable size in kB (4-?kb) + + This module supports multiple cards and autoprobe. + + The power-management is supported. + + Module snd-ua101 + ---------------- + + Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. + + This module supports multiple devices, autoprobe and hotplugging. + + Module snd-usb-audio + -------------------- + + Module for USB audio and USB MIDI devices. + + vid - Vendor ID for the device (optional) + pid - Product ID for the device (optional) + nrpacks - Max. number of packets per URB (default: 8) + device_setup - Device specific magic number (optional) + - Influence depends on the device + - Default: 0x0000 + ignore_ctl_error - Ignore any USB-controller regarding mixer + interface (default: no) + + This module supports multiple devices, autoprobe and hotplugging. + + NB: nrpacks parameter can be modified dynamically via sysfs. + Don't put the value over 20. Changing via sysfs has no sanity + check. + NB: ignore_ctl_error=1 may help when you get an error at accessing + the mixer element such as URB error -22. This happens on some + buggy USB device or the controller. + + Module snd-usb-caiaq + -------------------- + + Module for caiaq UB audio interfaces, + * Native Instruments RigKontrol2 + * Native Instruments Kore Controller + * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 8 DJ + + This module supports multiple devices, autoprobe and hotplugging. + + Module snd-usb-usx2y + -------------------- + + Module for Tascam USB US-122, US-224 and US-428 devices. + + This module supports multiple devices, autoprobe and hotplugging. + + Note: you need to load the firmware via usx2yloader utility included + in alsa-tools and alsa-firmware packages. + + Module snd-via82xx + ------------------ + + Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, + 8233A, 8233C, 8235, 8237 (south) bridge. + + mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup + [VIA686A/686B only] + joystick - Enable joystick (default off) [VIA686A/686B only] + ac97_clock - AC'97 codec clock base (default 48000Hz) + dxs_support - support DXS channels, + 0 = auto (default), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA, 5 = enable any sample + rate and different sample rates on different + channels + [VIA8233/C, 8235, 8237 only] + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + + This module supports one chip and autoprobe. + + Note: on some SMP motherboards like MSI 694D the interrupts might + not be generated properly. In such a case, please try to + set the SMP (or MPS) version on BIOS to 1.1 instead of + default value 1.4. Then the interrupt number will be + assigned under 15. You might also upgrade your BIOS. + + Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) + channels as the first PCM. On these channels, up to 4 + streams can be played at the same time, and the controller + can perform sample rate conversion with separate rates for + each channel. + As default (dxs_support = 0), 48k fixed rate is chosen + except for the known devices since the output is often + noisy except for 48k on some mother boards due to the + bug of BIOS. + Please try once dxs_support=5 and if it works on other + sample rates (e.g. 44.1kHz of mp3 playback), please let us + know the PCI subsystem vendor/device id's (output of + "lspci -nv"). + If dxs_support=5 does not work, try dxs_support=4; if it + doesn't work too, try dxs_support=1. (dxs_support=1 is + usually for old motherboards. The correct implemented + board should work with 4 or 5.) If it still doesn't + work and the default setting is ok, dxs_support=3 is the + right choice. If the default setting doesn't work at all, + try dxs_support=2 to disable the DXS channels. + In any cases, please let us know the result and the + subsystem vendor/device ids. See "Links and Addresses" + below. + + Note: for the MPU401 on VIA823x, use snd-mpu401 driver + additionally. The mpu_port option is for VIA686 chips only. + + The power-management is supported. + + Module snd-via82xx-modem + ------------------------ + + Module for VIA82xx AC97 modem + + ac97_clock - AC'97 codec clock base (default 48000Hz) + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-virmidi + ------------------ + + Module for virtual rawmidi devices. + This module creates virtual rawmidi devices which communicate + to the corresponding ALSA sequencer ports. + + midi_devs - MIDI devices # (1-4, default=4) + + This module supports multiple cards. + + Module snd-virtuoso + ------------------- + + Module for sound cards based on the Asus AV66/AV100/AV200 chips, + i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), + Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. + + This module supports autoprobe and multiple cards. + + Module snd-vx222 + ---------------- + + Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. + + mic - Enable Microphone on V222 Mic (NYI) + ibl - Capture IBL size. (default = 0, minimum size) + + This module supports multiple cards. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via vxloader utility in alsa-tools package. To invoke + vxloader automatically, add the following to /etc/modprobe.d/alsa.conf + + install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader + + (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to + /etc/modules.conf, instead.) + IBL size defines the interrupts period for PCM. The smaller size + gives smaller latency but leads to more CPU consumption, too. + The size is usually aligned to 126. As default (=0), the smallest + size is chosen. The possible IBL values can be found in + /proc/asound/cardX/vx-status proc file. + + The power-management is supported. + + Module snd-vxpocket + ------------------- + + Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. + + ibl - Capture IBL size. (default = 0, minimum size) + + This module supports multiple cards. The module is compiled only when + PCMCIA is supported on kernel. + + With the older 2.6.x kernel, to activate the driver via the card + manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the + sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no + longer require a config file. + + When the driver is compiled as a module and the hotplug firmware + is supported, the firmware data is loaded via hotplug automatically. + Install the necessary firmware files in alsa-firmware package. + When no hotplug fw loader is available, you need to load the + firmware via vxloader utility in alsa-tools package. + + About capture IBL, see the description of snd-vx222 module. + + Note: snd-vxp440 driver is merged to snd-vxpocket driver since + ALSA 1.0.10. + + The power-management is supported. + + Module snd-ymfpci + ----------------- + + Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). + + mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default, + 1 (auto-detect for YMF744/754 only) + fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default + 1 (auto-detect for YMF744/754 only) + joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default, + 1 (auto-detect) + rear_switch - enable shared rear/line-in switch (bool) + + This module supports autoprobe and multiple chips. + + The power-management is supported. + + Module snd-pdaudiocf + -------------------- + + Module for Sound Core PDAudioCF sound card. + + The power-management is supported. + + +AC97 Quirk Option +================= + +The ac97_quirk option is used to enable/override the workaround for +specific devices on drivers for on-board AC'97 controllers like +snd-intel8x0. Some hardware have swapped output pins between Master +and Headphone, or Surround (thanks to confusion of AC'97 +specifications from version to version :-) + +The driver provides the auto-detection of known problematic devices, +but some might be unknown or wrongly detected. In such a case, pass +the proper value with this option. + +The following strings are accepted: + - default Don't override the default setting + - none Disable the quirk + - hp_only Bind Master and Headphone controls as a single control + - swap_hp Swap headphone and master controls + - swap_surround Swap master and surround controls + - ad_sharing For AD1985, turn on OMS bit and use headphone + - alc_jack For ALC65x, turn on the jack sense mode + - inv_eapd Inverted EAPD implementation + - mute_led Bind EAPD bit for turning on/off mute LED + +For backward compatibility, the corresponding integer value -1, 0, +... are accepted, too. + +For example, if "Master" volume control has no effect on your device +but only "Headphone" does, pass ac97_quirk=hp_only module option. + + +Configuring Non-ISAPNP Cards +============================ + +When the kernel is configured with ISA-PnP support, the modules +supporting the isapnp cards will have module options "isapnp". +If this option is set, *only* the ISA-PnP devices will be probed. +For probing the non ISA-PnP cards, you have to pass "isapnp=0" option +together with the proper i/o and irq configuration. + +When the kernel is configured without ISA-PnP support, isapnp option +will be not built in. + + +Module Autoloading Support +========================== + +The ALSA drivers can be loaded automatically on demand by defining +module aliases. The string 'snd-card-%1' is requested for ALSA native +devices where %i is sound card number from zero to seven. + +To auto-load an ALSA driver for OSS services, define the string +'sound-slot-%i' where %i means the slot number for OSS, which +corresponds to the card index of ALSA. Usually, define this +as the same card module. + +An example configuration for a single emu10k1 card is like below: +----- /etc/modprobe.d/alsa.conf +alias snd-card-0 snd-emu10k1 +alias sound-slot-0 snd-emu10k1 +----- /etc/modprobe.d/alsa.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.d/alsa.conf +# ALSA portion +options snd cards_limit=2 +alias snd-card-0 snd-interwave +alias snd-card-1 snd-ens1371 +options snd-interwave index=0 +options snd-ens1371 index=1 +# OSS/Free portion +alias sound-slot-0 snd-interwave +alias sound-slot-1 snd-ens1371 +----- /etc/modprobe.d/alsa.conf + +In this example, the interwave card is always loaded as the first card +(index 0) and ens1371 as the second (index 1). + +Alternative (and new) way to fixate the slot assignment is to use +"slots" option of snd module. In the case above, specify like the +following: + +options snd slots=snd-interwave,snd-ens1371 + +Then, the first slot (#0) is reserved for snd-interwave driver, and +the second (#1) for snd-ens1371. You can omit index option in each +driver if slots option is used (although you can still have them at +the same time as long as they don't conflict). + +The slots option is especially useful for avoiding the possible +hot-plugging and the resultant slot conflict. For example, in the +case above again, the first two slots are already reserved. If any +other driver (e.g. snd-usb-audio) is loaded before snd-interwave or +snd-ens1371, it will be assigned to the third or later slot. + +When a module name is given with '!', the slot will be given for any +modules but that name. For example, "slots=!snd-pcsp" will reserve +the first slot for any modules but snd-pcsp. + + +ALSA PCM devices to OSS devices mapping +======================================= + +/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 +/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 +/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 +/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 +/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 +/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 +/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 +/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 +/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 + +The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means +sound card number and second means device number. The ALSA devices +have either 'c' or 'p' suffix indicating the direction, capture and +playback, respectively. + +Please note that the device mapping above may be varied via the module +options of snd-pcm-oss module. + + +Proc interfaces (/proc/asound) +============================== + +/proc/asound/card#/pcm#[cp]/oss +------------------------------- + String "erase" - erase all additional information about OSS applications + String "<app_name> <fragments> <fragment_size> [<options>]" + + <app_name> - name of application with (higher priority) or without path + <fragments> - number of fragments or zero if auto + <fragment_size> - size of fragment in bytes or zero if auto + <options> - optional parameters + - disable the application tries to open a pcm device for + this channel but does not want to use it. + (Cause a bug or mmap needs) + It's good for Quake etc... + - direct don't use plugins + - block force block mode (rvplayer) + - non-block force non-block mode + - whole-frag write only whole fragments (optimization affecting + playback only) + - no-silence do not fill silence ahead to avoid clicks + - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl + instead of filled blocks + + Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss + echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss + echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss + + +Early Buffer Allocation +======================= + +Some drivers (e.g. hdsp) require the large contiguous buffers, and +sometimes it's too late to find such spaces when the driver module is +actually loaded due to memory fragmentation. You can pre-allocate the +PCM buffers by loading snd-page-alloc module and write commands to its +proc file in prior, for example, in the early boot stage like +/etc/init.d/*.local scripts. + +Reading the proc file /proc/drivers/snd-page-alloc shows the current +usage of page allocation. In writing, you can send the following +commands to the snd-page-alloc driver: + + - add VENDOR DEVICE MASK SIZE BUFFERS + + VENDOR and DEVICE are PCI vendor and device IDs. They take + integer numbers (0x prefix is needed for the hex). + MASK is the PCI DMA mask. Pass 0 if not restricted. + SIZE is the size of each buffer to allocate. You can pass + k and m suffix for KB and MB. The max number is 16MB. + BUFFERS is the number of buffers to allocate. It must be greater + than 0. The max number is 4. + + - erase + + This will erase the all pre-allocated buffers which are not in + use. + + +Links and Addresses +=================== + + ALSA project homepage + http://www.alsa-project.org + + Kernel Bugzilla + http://bugzilla.kernel.org/ + + ALSA Developers ML + mailto:alsa-devel@alsa-project.org + + alsa-info.sh script + http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt new file mode 100644 index 000000000..7f10dc6ff --- /dev/null +++ b/Documentation/sound/alsa/Audigy-mixer.txt @@ -0,0 +1,345 @@ + + Sound Blaster Audigy mixer / default DSP code + =========================================== + +This is based on SB-Live-mixer.txt. + +The EMU10K2 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K2 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + +name='PCM Front Playback Volume',index=0 + +This control is used to attenuate samples for left and right front PCM FX-bus +accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM +samples for 5.1 playback. The result samples are forwarded to the front DAC PCM +slots of the Philips DAC. + +name='PCM Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right surround PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM +samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM +slots of the Philips DAC. + +name='PCM Center Playback Volume',index=0 + +This control is used to attenuate samples for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample +is forwarded to the center DAC PCM slot of the Philips DAC. + +name='PCM LFE Playback Volume',index=0 + +This control is used to attenuate sample for LFE PCM FX-bus accumulator. +ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample +is forwarded to the LFE DAC PCM slot of the Philips DAC. + +name='PCM Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for +stereo playback. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='PCM Capture Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Music Playback Volume',index=0 + +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Music Capture Volume',index=0 + +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Mic Playback Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +For Mic input is used AC97 codec. The result samples are forwarded to +the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic +capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. + +name='Mic Capture Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Audigy CD Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the Philips DAC. + +name='Audigy CD Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 Optical Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='IEC958 Optical Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Line2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Analog Mix Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs from Philips ADC. The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. This contains mix from analog sources +like CD, Line In, Aux, .... + +name='Analog Mix Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs Philips ADC. The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Aux2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Aux2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Front Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right front speakers of +this mix. + +name='Surround Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right surround speakers of +this mix. + +name='Center Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for center speaker of this mix. + +name='LFE Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for LFE speaker of this mix. + +name='Tone Control - Switch',index=0 + +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 + +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 + +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Master Playback Volume',index=0 + +This control is used to attenuate samples for front, surround, center and +LFE outputs. + +name='IEC958 Optical Raw Playback Switch',index=0 + +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + + +2) PCM stream related controls +------------------------------ + +name='EMU10K1 PCM Volume',index 0-31 + +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + + 0 - mono, default 0xffff (no attenuation) + 1 - left, default 0xffff (no attenuation) + 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 + +This control specifies the destination - FX-bus accumulators. There 24 +values with this mapping: + + 0 - mono, A destination (FX-bus 0-63), default 0 + 1 - mono, B destination (FX-bus 0-63), default 1 + 2 - mono, C destination (FX-bus 0-63), default 2 + 3 - mono, D destination (FX-bus 0-63), default 3 + 4 - mono, E destination (FX-bus 0-63), default 0 + 5 - mono, F destination (FX-bus 0-63), default 0 + 6 - mono, G destination (FX-bus 0-63), default 0 + 7 - mono, H destination (FX-bus 0-63), default 0 + 8 - left, A destination (FX-bus 0-63), default 0 + 9 - left, B destination (FX-bus 0-63), default 1 + 10 - left, C destination (FX-bus 0-63), default 2 + 11 - left, D destination (FX-bus 0-63), default 3 + 12 - left, E destination (FX-bus 0-63), default 0 + 13 - left, F destination (FX-bus 0-63), default 0 + 14 - left, G destination (FX-bus 0-63), default 0 + 15 - left, H destination (FX-bus 0-63), default 0 + 16 - right, A destination (FX-bus 0-63), default 0 + 17 - right, B destination (FX-bus 0-63), default 1 + 18 - right, C destination (FX-bus 0-63), default 2 + 19 - right, D destination (FX-bus 0-63), default 3 + 20 - right, E destination (FX-bus 0-63), default 0 + 21 - right, F destination (FX-bus 0-63), default 0 + 22 - right, G destination (FX-bus 0-63), default 0 + 23 - right, H destination (FX-bus 0-63), default 0 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 + +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + + 0 - mono, A destination attn, default 255 (no attenuation) + 1 - mono, B destination attn, default 255 (no attenuation) + 2 - mono, C destination attn, default 0 (mute) + 3 - mono, D destination attn, default 0 (mute) + 4 - mono, E destination attn, default 0 (mute) + 5 - mono, F destination attn, default 0 (mute) + 6 - mono, G destination attn, default 0 (mute) + 7 - mono, H destination attn, default 0 (mute) + 8 - left, A destination attn, default 255 (no attenuation) + 9 - left, B destination attn, default 0 (mute) + 10 - left, C destination attn, default 0 (mute) + 11 - left, D destination attn, default 0 (mute) + 12 - left, E destination attn, default 0 (mute) + 13 - left, F destination attn, default 0 (mute) + 14 - left, G destination attn, default 0 (mute) + 15 - left, H destination attn, default 0 (mute) + 16 - right, A destination attn, default 0 (mute) + 17 - right, B destination attn, default 255 (no attenuation) + 18 - right, C destination attn, default 0 (mute) + 19 - right, D destination attn, default 0 (mute) + 20 - right, E destination attn, default 0 (mute) + 21 - right, F destination attn, default 0 (mute) + 22 - right, G destination attn, default 0 (mute) + 23 - right, H destination attn, default 0 (mute) + + + +4) MANUALS/PATENTS: +------------------- + +ftp://opensource.creative.com/pub/doc +------------------------------------- + + Files: + LM4545.pdf AC97 Codec + + m2049.pdf The EMU10K1 Digital Audio Processor + + hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + Patent numbers: + WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) + streams + + WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + + WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + + US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + + US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + + US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + + US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) + + US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + + US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) + pool of short term memory registers + + US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 000000000..e7a5ed4dc --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,442 @@ + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 + ======================================================== + + Thibault Le Meur <Thibault.LeMeur@supelec.fr> + +This document is a guide to using the M-Audio Audiophile USB (tm) device with +ALSA and JACK. + +History +======= +* v1.4 - Thibault Le Meur (2007-07-11) + - Added Low Endianness nature of 16bits-modes + found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> + - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + +1 - Audiophile USB Specs and correct usage +========================================== + +This part is a reminder of important facts about the functions and limitations +of the device. + +The device has 4 audio interfaces, and 2 MIDI ports: + * Analog Stereo Input (Ai) + - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) + - When the 1/4" TS (jack) connectors are connected, the RCA connectors + are disabled + * Analog Stereo Output (Ao) + * Digital Stereo Input (Di) + * Digital Stereo Output (Do) + * Midi In (Mi) + * Midi Out (Mo) + +The internal DAC/ADC has the following characteristics: +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" + +Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be +activated at the same time depending on the audio mode selected: + * 16-bit/48kHz ==> 4 channels in + 4 channels out + - Ai+Ao+Di+Do + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out + - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) + - Ai or Ao or Di or Do + +Important facts about the Digital interface: +-------------------------------------------- + * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under Linux + - Note that in this setup only the Do interface can be enabled + * Apart from recording an audio digital stream, enabling the Di port is a way +to synchronize the device to an external sample clock + - As a consequence, the Di port must be enable only if an active Digital +source is connected + - Enabling Di when no digital source is connected can result in a +synchronization error (for instance sound played at an odd sample rate) + + +2 - Audiophile USB MIDI support in ALSA +======================================= + +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + * snd-usb-audio + * snd-seq-midi + +No additional setting is required. + + +3 - Audiophile USB Audio support in ALSA +======================================== + +Audio functions of the Audiophile USB device are handled by the snd-usb-audio +module. This module can work in a default mode (without any device-specific +parameter), or in an "advanced" mode with the device-specific parameter called +"device_setup". + +3.1 - Default Alsa driver mode +------------------------------ + +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): + * hw:1,0 is Ao in playback and Di in capture + * hw:1,1 is Do in playback and Ai in capture + * hw:1,2 is Do in AC3/DTS passthrough mode + +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. + +Examples: + * playing a S24_3BE encoded raw file to the Ao port + % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw + * recording a S24_3BE encoded raw file from the Ai port + % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw + * playing a S16_BE encoded raw file to the Do port + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw + +If you're happy with the default Alsa driver mode and don't experience any +issue with this mode, then you can skip the following chapter. + +3.2 - Advanced module setup +--------------------------- + +Due to the hardware constraints described above, the device initialization made +by the Alsa driver in default mode may result in a corrupted state of the +device. For instance, a particularly annoying issue is that the sound captured +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). + +For people having this problem, the snd-usb-audio module has a new module +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) + +3.2.1 - Initializing the working mode of the Audiophile USB + +As far as the Audiophile USB device is concerned, this value let the user +specify: + * the sample depth + * the sample rate + * whether the Di port is used or not + +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + + * device_setup=0x01 + - 16bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x11 + - 16bits 48kHz mode with Di enabled + - Ai,Ao,Di,Do can be used at the same time + - hw:1,0 is available in capture mode + - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + + * device_setup=0x09 + - 24bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x19 + - 24bits 48kHz mode with Di enabled + - 3 ports from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in capture mode and an active digital source must be + connected to Di + - hw:1,2 is not available + + * device_setup=0x0D or 0x10 + - 24bits 96kHz mode + - Di is enabled by default for this mode but does not need to be connected + to an active source + - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in captured mode + - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + + * device_setup=0x03 + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +3.2.2 - How to use the device_setup parameter +---------------------------------------------- + +The parameter can be given: + + * By manually probing the device (as root): + # modprobe -r snd-usb-audio + # modprobe snd-usb-audio index=1 device_setup=0x09 + + * Or while configuring the modules options in your modules configuration file + (typically a .conf file in /etc/modprobe.d/ directory: + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +CAUTION when initializing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead to a misconfiguration of the device. In this case + turn off the device, unprobe the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: + - first turn off the device + - de-register the snd-usb-audio module (modprobe -r) + - change the device_setup parameter by changing the device_setup + option in /etc/modprobe.d/*.conf + - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. + +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure + +If you want to understand the device_setup magic numbers for the Audiophile +USB, you need some very basic understanding of binary computation. However, +this is not required to use the parameter and you may skip this section. + +The device_setup is one byte long and its structure is the following: + + +---+---+---+---+---+---+---+---+ + | b7| b6| b5| b4| b3| b2| b1| b0| + +---+---+---+---+---+---+---+---+ + | 0 | 0 | 0 | Di|24B|96K|DTS|SET| + +---+---+---+---+---+---+---+---+ + +Where: + * b0 is the "SET" bit + - it MUST be set if device_setup is initialized + * b1 is the "DTS" bit + - it is set only for Digital output with DTS/AC3 + - this setup is not tested + * b2 is the Rate selection flag + - When set to "1" the rate range is 48.1-96kHz + - Otherwise the sample rate range is 8-48kHz + * b3 is the bit depth selection flag + - When set to "1" samples are 24bits long + - Otherwise they are 16bits long + - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits + samples + * b4 is the Digital input flag + - When set to "1" the device assumes that an active digital source is + connected + - You shouldn't enable Di if no source is seen on the port (this leads to + synchronization issues) + - b4 is implied by b2 (since only one port is enabled at a time no synch + error can occur) + * b5 to b7 are reserved for future uses, and must be set to "0" + - might become Ao, Do, Ai, for b7, b6, b4 respectively + +Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since + b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll + only be able to use one at the same time + +3.2.3.2 - USB implementation details for this device + +You may safely skip this section if you're not interested in driver +hacking. + +This section describes some internal aspects of the device and summarizes the +data I got by usb-snooping the windows and Linux drivers. + +The M-Audio Audiophile USB has 7 USB Interfaces: +a "USB interface": + * USB Interface nb.0 + * USB Interface nb.1 + - Audio Control function + * USB Interface nb.2 + - Analog Output + * USB Interface nb.3 + - Digital Output + * USB Interface nb.4 + - Analog Input + * USB Interface nb.5 + - Digital Input + * USB Interface nb.6 + - MIDI interface compliant with the MIDIMAN quirk + +Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: + * Interface 3 (Digital Out) has an extra Alset nb.6 + * Interface 5 (Digital In) does not have Alset nb.3 and 5 + +Here is a short description of the AltSettings capabilities: + * AltSettings 1 corresponds to + - 24-bit depth, 48.1-96kHz sample mode + - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) + * AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 6 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch playback (Do), audio format type III IEC1937_AC-3 + +In order to ensure a correct initialization of the device, the driver +_must_know_ how the device will be used: + * if DTS is chosen, only Interface 2 with AltSet nb.6 must be + registered + * if 96KHz only AltSets nb.1 of each interface must be selected + * if samples are using 24bits/48KHz then AltSet 2 must me used if + Digital input is connected, and only AltSet nb.3 if Digital input + is not connected + * if samples are using 16bits/48KHz then AltSet 4 must me used if + Digital input is connected, and only AltSet nb.5 if Digital input + is not connected + +When device_setup is given as a parameter to the snd-usb-audio module, the +parse_audio_endpoints function uses a quirk called +"audiophile_skip_setting_quirk" in order to prevent AltSettings not +corresponding to device_setup from being registered in the driver. + +4 - Audiophile USB and Jack support +=================================== + +This section deals with support of the Audiophile USB device in Jack. + +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +4.1 - Direct support in Jackd +----------------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +exactly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. + +For instance here is one way to run Jack with 2 playback channels on Ao and 2 +capture channels from Ai: + % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 + +However you may see the following warning message: +"You appear to be using the ALSA software "plug" layer, probably a result of +using the "default" ALSA device. This is less efficient than it could be. +Consider using a hardware device instead rather than using the plug layer." + +4.3 - Getting 2 input and/or output interfaces in Jack +------------------------------------------------------ + +As you can see, starting the Jack server this way will only enable 1 stereo +input (Di or Ai) and 1 stereo output (Ao or Do). + +This is due to the following restrictions: +* Jack can only open one capture device and one playback device at a time +* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 + (and optionally hw:1,2) + +If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to +combine the Alsa devices into one logical "complex" device. + +If you want to give it a try, I recommend reading the information from +this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html +It is related to another device (ice1712) but can be adapted to suit +the Audiophile USB. + +Enabling multiple Audiophile USB interfaces for Jackd will certainly require: +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc + file +* start jackd with this device + +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt new file mode 100644 index 000000000..f158cde8b --- /dev/null +++ b/Documentation/sound/alsa/Bt87x.txt @@ -0,0 +1,78 @@ +Intro +===== + +You might have noticed that the bt878 grabber cards have actually +_two_ PCI functions: + +$ lspci +[ ... ] +00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) +00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02) +[ ... ] + +The first does video, it is backward compatible to the bt848. The second +does audio. snd-bt87x is a driver for the second function. It's a sound +driver which can be used for recording sound (and _only_ recording, no +playback). As most TV cards come with a short cable which can be plugged +into your sound card's line-in you probably don't need this driver if all +you want to do is just watching TV... + +Some cards do not bother to connect anything to the audio input pins of +the chip, and some other cards use the audio function to transport MPEG +video data, so it's quite possible that audio recording may not work +with your card. + + +Driver Status +============= + +The driver is now stable. However, it doesn't know about many TV cards, +and it refuses to load for cards it doesn't know. + +If the driver complains ("Unknown TV card found, the audio driver will +not load"), you can specify the load_all=1 option to force the driver to +try to use the audio capture function of your card. If the frequency of +recorded data is not right, try to specify the digital_rate option with +other values than the default 32000 (often it's 44100 or 64000). + +If you have an unknown card, please mail the ID and board name to +<alsa-devel@alsa-project.org>, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. + + +Audio modes +=========== + +The chip knows two different modes (digital/analog). snd-bt87x +registers two PCM devices, one for each mode. They cannot be used at +the same time. + + +Digital audio mode +================== + +The first device (hw:X,0) gives you 16 bit stereo sound. The sample +rate depends on the external source which feeds the Bt87x with digital +sound via I2S interface. + + +Analog audio mode (A/D) +======================= + +The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported +sample rates are between 119466 and 448000 Hz (yes, these numbers are +that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the +maximum sample rate is 1792000 Hz, but audio data becomes unusable +beyond 896000 Hz on my card. + +The chip has three analog inputs. Consequently you'll get a mixer +device to control these. + + +Have fun, + + Clemens + + +Written by Clemens Ladisch <clemens@ladisch.de> +big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org> diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt new file mode 100644 index 000000000..4e36e6e80 --- /dev/null +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -0,0 +1,254 @@ + Brief Notes on C-Media 8338/8738/8768/8770 Driver + ================================================= + + Takashi Iwai <tiwai@suse.de> + + +Front/Rear Multi-channel Playback +--------------------------------- + +CM8x38 chip can use ADC as the second DAC so that two different stereo +channels can be used for front/rear playbacks. Since there are two +DACs, both streams are handled independently unlike the 4/6ch multi- +channel playbacks in the section below. + +As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for +card#0) for front and 4/6ch playbacks, while the second PCM device +(hw:0,1) is assigned to the second DAC for rear playback. + +There are slight differences between the two DACs: + +- The first DAC supports U8 and S16LE formats, while the second DAC + supports only S16LE. +- The second DAC supports only two channel stereo. + +Please note that the CM8x38 DAC doesn't support continuous playback +rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000, +44100 and 48000 Hz. + +The rear output can be heard only when "Four Channel Mode" switch is +disabled. Otherwise no signal will be routed to the rear speakers. +As default it's turned on. + +*** WARNING *** +When "Four Channel Mode" switch is off, the output from rear speakers +will be FULL VOLUME regardless of Master and PCM volumes. +This might damage your audio equipment. Please disconnect speakers +before your turn off this switch. +*** WARNING *** + +[ Well.. I once got the output with correct volume (i.e. same with the + front one) and was so excited. It was even with "Four Channel" bit + on and "double DAC" mode. Actually I could hear separate 4 channels + from front and rear speakers! But.. after reboot, all was gone. + It's a very pity that I didn't save the register dump at that + time.. Maybe there is an unknown register to achieve this... ] + +If your card has an extra output jack for the rear output, the rear +playback should be routed there as default. If not, there is a +control switch in the driver "Line-In As Rear", which you can change +via alsamixer or somewhat else. When this switch is on, line-in jack +is used as rear output. + +There are two more controls regarding to the rear output. +The "Exchange DAC" switch is used to exchange front and rear playback +routes, i.e. the 2nd DAC is output from front output. + + +4/6 Multi-Channel Playback +-------------------------- + +The recent CM8738 chips support for the 4/6 multi-channel playback +function. This is useful especially for AC3 decoding. + +When the multi-channel is supported, the driver name has a suffix +"-MC" such like "CMI8738-MC6". You can check this name from +/proc/asound/cards. + +When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or +4) channels. While the dual DAC supports two different rates or +formats, the 4/6-ch playback supports only the same condition for all +channels. Since the multi-channel playback mode uses both DACs, you +cannot operate with full-duplex. + +The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" +in alsa-lib. For example, you can play a WAV file with 6 channels like + + % aplay -Dsurround51 sixchannels.wav + +For programming the 4/6 channel playback, you need to specify the PCM +channels as you like and set the format S16LE. For example, for playback +with 4 channels, + + snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); + // or mmap if you like + snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE); + snd_pcm_hw_params_set_channels(pcm, hw, 4); + +and use the interleaved 4 channel data. + +There are some control switches affecting to the speaker connections: + +"Line-In Mode" - an enum control to change the behavior of line-in + jack. Either "Line-In", "Rear Output" or "Bass Output" can + be selected. The last item is available only with model 039 + or newer. + When "Rear Output" is chosen, the surround channels 3 and 4 + are output to line-in jack. +"Mic-In Mode" - an enum control to change the behavior of mic-in + jack. Either "Mic-In" or "Center/LFE Output" can be + selected. + When "Center/LFE Output" is chosen, the center and bass + channels (channels 5 and 6) are output to mic-in jack. + +Digital I/O +----------- + +The CM8x38 provides the excellent SPDIF capability with very cheap +price (yes, that's the reason I bought the card :) + +The SPDIF playback and capture are done via the third PCM device +(hw:0,2). Usually this is assigned to the PCM device "spdif". +The available rates are 44100 and 48000 Hz. +For playback with aplay, you can run like below: + + % aplay -Dhw:0,2 foo.wav + +or + + % aplay -Dspdif foo.wav + +24bit format is also supported experimentally. + +The playback and capture over SPDIF use normal DAC and ADC, +respectively, so you cannot playback both analog and digital streams +simultaneously. + +To enable SPDIF output, you need to turn on "IEC958 Output Switch" +control via mixer or alsactl ("IEC958" is the official name of +so-called S/PDIF). Then you'll see the red light on from the card so +you know that's working obviously :) +The SPDIF input is always enabled, so you can hear SPDIF input data +from line-out with "IEC958 In Monitor" switch at any time (see +below). + +You can play via SPDIF even with the first device (hw:0,0), +but SPDIF is enabled only when the proper format (S16LE), sample rate +(441100 or 48000) and channels (2) are used. Otherwise it's turned +off. (Also don't forget to turn on "IEC958 Output Switch", too.) + + +Additionally there are relevant control switches: + +"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and + output through SPDIF. This switch appears only on old chip + models (CM8738 033 and 037). + Note: without this control you can output PCM to SPDIF. + This is "mixing" of streams, so e.g. it's not for AC3 output + (see the next section). + +"IEC958 In Select" - Select SPDIF input, the internal CD-in (false) + and the external input (true). + +"IEC958 Loop" - SPDIF input data is loop back into SPDIF + output (aka bypass) + +"IEC958 Copyright" - Set the copyright bit. + +"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface. + On some cards this doesn't work and you need to change the + configuration with hardware dip-switch. + +"IEC958 In Monitor" - SPDIF input is routed to DAC. + +"IEC958 In Phase Inverse" - Set SPDIF input format as inverse. + [FIXME: this doesn't work on all chips..] + +"IEC958 In Valid" - Set input validity flag detection. + +Note: When "PCM Playback Switch" is on, you'll hear the digital output +stream through analog line-out. + + +The AC3 (RAW DIGITAL) OUTPUT +---------------------------- + +The driver supports raw digital (typically AC3) i/o over SPDIF. This +can be toggled via IEC958 playback control, but usually you need to +access it via alsa-lib. See alsa-lib documents for more details. + +On the raw digital mode, the "PCM Playback Switch" is automatically +turned off so that non-audio data is heard from the analog line-out. +Similarly the following switches are off: "IEC958 Mix Analog" and +"IEC958 Loop". The switches are resumed after closing the SPDIF PCM +device automatically to the previous state. + +On the model 033, AC3 is implemented by the software conversion in +the alsa-lib. If you need to bypass the software conversion of IEC958 +subframes, pass the "soft_ac3=0" module option. This doesn't matter +on the newer models. + + +ANALOG MIXER INTERFACE +---------------------- + +The mixer interface on CM8x38 is similar to SB16. +There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback +volumes. Synth, CD, Line and Mic have playback and capture switches, +too, as well as SB16. + +In addition to the standard SB mixer, CM8x38 provides more functions. +- PCM playback switch +- PCM capture switch (to capture the data sent to DAC) +- Mic Boost switch +- Mic capture volume +- Aux playback volume/switch and capture switch +- 3D control switch + + +MIDI CONTROLLER +--------------- + +With CMI8338 chips, the MPU401-UART interface is disabled as default. +You need to set the module option "mpu_port" to a valid I/O port address +to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and +0x330. Choose a value that doesn't conflict with other cards. + +With CMI8738 and newer chips, the MIDI interface is enabled by default +and the driver automatically chooses a port address. + +There is _no_ hardware wavetable function on this chip (except for +OPL3 synth below). +What's said as MIDI synth on Windows is a software synthesizer +emulation. On Linux use TiMidity or other softsynth program for +playing MIDI music. + + +FM OPL/3 Synth +-------------- + +The FM OPL/3 is also enabled as default only for the first card. +Set "fm_port" module option for more cards. + +The output quality of FM OPL/3 is, however, very weird. +I don't know why.. + +CMI8768 and newer chips do not have the FM synth. + + +Joystick and Modem +------------------ + +The legacy joystick is supported. To enable the joystick support, pass +joystick_port=1 module option. The value 1 means the auto-detection. +If the auto-detection fails, try to pass the exact I/O address. + +The modem is enabled dynamically via a card control switch "Modem". + + +Debugging Information +--------------------- + +The registers are shown in /proc/asound/cardX/cmipci. If you have any +problem (especially unexpected behavior of mixer), please attach the +output of this proc file together with the bug report. diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt new file mode 100644 index 000000000..3c43d1a4c --- /dev/null +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -0,0 +1,153 @@ +ALSA PCM channel-mapping API +============================ + Takashi Iwai <tiwai@suse.de> + +GENERAL +------- + +The channel mapping API allows user to query the possible channel maps +and the current channel map, also optionally to modify the channel map +of the current stream. + +A channel map is an array of position for each PCM channel. +Typically, a stereo PCM stream has a channel map of + { front_left, front_right } +while a 4.0 surround PCM stream has a channel map of + { front left, front right, rear left, rear right }. + +The problem, so far, was that we had no standard channel map +explicitly, and applications had no way to know which channel +corresponds to which (speaker) position. Thus, applications applied +wrong channels for 5.1 outputs, and you hear suddenly strange sound +from rear. Or, some devices secretly assume that center/LFE is the +third/fourth channels while others that C/LFE as 5th/6th channels. + +Also, some devices such as HDMI are configurable for different speaker +positions even with the same number of total channels. However, there +was no way to specify this because of lack of channel map +specification. These are the main motivations for the new channel +mapping API. + + +DESIGN +------ + +Actually, "the channel mapping API" doesn't introduce anything new in +the kernel/user-space ABI perspective. It uses only the existing +control element features. + +As a ground design, each PCM substream may contain a control element +providing the channel mapping information and configuration. This +element is specified by: + iface = SNDRV_CTL_ELEM_IFACE_PCM + name = "Playback Channel Map" or "Capture Channel Map" + device = the same device number for the assigned PCM substream + index = the same index number for the assigned PCM substream + +Note the name is different depending on the PCM substream direction. + +Each control element provides at least the TLV read operation and the +read operation. Optionally, the write operation can be provided to +allow user to change the channel map dynamically. + +* TLV + +The TLV operation gives the list of available channel +maps. A list item of a channel map is usually a TLV of + type data-bytes ch0 ch1 ch2... +where type is the TLV type value, the second argument is the total +bytes (not the numbers) of channel values, and the rest are the +position value for each channel. + +As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED, +SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used. +The _FIXED type is for a channel map with the fixed channel position +while the latter two are for flexible channel positions. _VAR type is +for a channel map where all channels are freely swappable and _PAIRED +type is where pair-wise channels are swappable. For example, when you +have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap +only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and +RR. + +These new TLV types are defined in sound/tlv.h. + +The available channel position values are defined in sound/asound.h, +here is a cut: + +/* channel positions */ +enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_RC, /* rear center */ + /* new definitions */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, +}; + +When a PCM stream can provide more than one channel map, you can +provide multiple channel maps in a TLV container type. The TLV data +to be returned will contain such as: + SNDRV_CTL_TLVT_CONTAINER 96 + SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC + SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR + SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \ + SNDRV_CHMAP_RL SNDRV_CHMAP_RR + +The channel position is provided in LSB 16bits. The upper bits are +used for bit flags. + +#define SNDRV_CHMAP_POSITION_MASK 0xffff +#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) +#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + +SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted, +(thus summing left and right channels would result in almost silence). +Some digital mic devices have this. + +When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values +don't follow the standard definition above but driver-specific. + +* READ OPERATION + +The control read operation is for providing the current channel map of +the given stream. The control element returns an integer array +containing the position of each channel. + +When this is performed before the number of the channel is specified +(i.e. hw_params is set), it should return all channels set to +UNKNOWN. + +* WRITE OPERATION + +The control write operation is optional, and only for devices that can +change the channel configuration on the fly, such as HDMI. User needs +to pass an integer value containing the valid channel positions for +all channels of the assigned PCM substream. + +This operation is allowed only at PCM PREPARED state. When called in +other states, it shall return an error. diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt new file mode 100644 index 000000000..3fc1cf50d --- /dev/null +++ b/Documentation/sound/alsa/ControlNames.txt @@ -0,0 +1,107 @@ +This document describes standard names of mixer controls. + +Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION + +DIRECTION: + <nothing> (both directions) + Playback + Capture + Bypass Playback + Bypass Capture + +FUNCTION: + Switch (on/off switch) + Volume + Route (route control, hardware specific) + +CHANNEL: + <nothing> (channel independent, or applies to all channels) + Front + Surround (rear left/right in 4.0/5.1 surround) + CLFE + Center + LFE + Side (side left/right for 7.1 surround) + +LOCATION: (physical location of source) + Front + Rear + Dock (docking station) + Internal + +SOURCE: + Master + Master Mono + Hardware Master + Speaker (internal speaker) + Bass Speaker (internal LFE speaker) + Headphone + Line Out + Beep (beep generator) + Phone + Phone Input + Phone Output + Synth + FM + Mic + Headset Mic (mic part of combined headset jack - 4-pin headphone + mic) + Headphone Mic (mic part of either/or - 3-pin headphone or mic) + Line (input only, use "Line Out" for output) + CD + Video + Zoom Video + Aux + PCM + 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 + HDMI + SPDIF (output only) + SPDIF In + Digital In + HDMI/DP (either HDMI or DisplayPort) + +Exceptions (deprecated): + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|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/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt new file mode 100644 index 000000000..e9621e349 --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio-Controls.txt @@ -0,0 +1,116 @@ +This file explains the codec-specific mixer controls. + +Realtek codecs +-------------- + +* Channel Mode + This is an enum control to change the surround-channel setup, + appears only when the surround channels are available. + It gives the number of channels to be used, "2ch", "4ch", "6ch", + and "8ch". According to the configuration, this also controls the + jack-retasking of multi-I/O jacks. + +* Auto-Mute Mode + This is an enum control to change the auto-mute behavior of the + headphone and line-out jacks. If built-in speakers and headphone + and/or line-out jacks are available on a machine, this controls + appears. + When there are only either headphones or line-out jacks, it gives + "Disabled" and "Enabled" state. When enabled, the speaker is muted + automatically when a jack is plugged. + + When both headphone and line-out jacks are present, it gives + "Disabled", "Speaker Only" and "Line-Out+Speaker". When + speaker-only is chosen, plugging into a headphone or a line-out jack + mutes the speakers, but not line-outs. When line-out+speaker is + selected, plugging to a headphone jack mutes both speakers and + line-outs. + + +IDT/Sigmatel codecs +------------------- + +* Analog Loopback + This control enables/disables the analog-loopback circuit. This + appears only when "loopback" is set to true in a codec hint + (see HD-Audio.txt). Note that on some codecs the analog-loopback + and the normal PCM playback are exclusive, i.e. when this is on, you + won't hear any PCM stream. + +* Swap Center/LFE + Swaps the center and LFE channel order. Normally, the left + corresponds to the center and the right to the LFE. When this is + ON, the left to the LFE and the right to the center. + +* Headphone as Line Out + When this control is ON, treat the headphone jacks as line-out + jacks. That is, the headphone won't auto-mute the other line-outs, + and no HP-amp is set to the pins. + +* Mic Jack Mode, Line Jack Mode, etc + These enum controls the direction and the bias of the input jack + pins. Depending on the jack type, it can set as "Mic In" and "Line + In", for determining the input bias, or it can be set to "Line Out" + when the pin is a multi-I/O jack for surround channels. + + +VIA codecs +---------- + +* Smart 5.1 + An enum control to re-task the multi-I/O jacks for surround outputs. + When it's ON, the corresponding input jacks (usually a line-in and a + mic-in) are switched as the surround and the CLFE output jacks. + +* Independent HP + When this enum control is enabled, the headphone output is routed + from an individual stream (the third PCM such as hw:0,2) instead of + the primary stream. In the case the headphone DAC is shared with a + side or a CLFE-channel DAC, the DAC is switched to the headphone + automatically. + +* Loopback Mixing + An enum control to determine whether the analog-loopback route is + enabled or not. When it's enabled, the analog-loopback is mixed to + the front-channel. Also, the same route is used for the headphone + and speaker outputs. As a side-effect, when this mode is set, the + individual volume controls will be no longer available for + headphones and speakers because there is only one DAC connected to a + mixer widget. + +* Dynamic Power-Control + This control determines whether the dynamic power-control per jack + detection is enabled or not. When enabled, the widgets power state + (D0/D3) are changed dynamically depending on the jack plugging + state for saving power consumptions. However, if your system + doesn't provide a proper jack-detection, this won't work; in such a + case, turn this control OFF. + +* Jack Detect + This control is provided only for VT1708 codec which gives no proper + unsolicited event per jack plug. When this is on, the driver polls + the jack detection so that the headphone auto-mute can work, while + turning this off would reduce the power consumption. + + +Conexant codecs +--------------- + +* Auto-Mute Mode + See Reatek codecs. + + +Analog codecs +-------------- + +* Channel Mode + This is an enum control to change the surround-channel setup, + appears only when the surround channels are available. + It gives the number of channels to be used, "2ch", "4ch" and "6ch". + According to the configuration, this also controls the + jack-retasking of multi-I/O jacks. + +* Independent HP + When this enum control is enabled, the headphone output is routed + from an individual stream (the third PCM such as hw:0,2) instead of + the primary stream. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt new file mode 100644 index 000000000..5a3163cac --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -0,0 +1,314 @@ + Model name Description + ---------- ----------- +ALC880 +====== + 3stack 3-jack in back and a headphone out + 3stack-digout 3-jack in back, a HP out and a SPDIF out + 5stack 5-jack in back, 2-jack in front + 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out + 6stack 6-jack in back, 2-jack in front + 6stack-digout 6-jack with a SPDIF out + +ALC260 +====== + N/A + +ALC262 +====== + inv-dmic Inverted internal mic workaround + +ALC267/268 +========== + inv-dmic Inverted internal mic workaround + +ALC269/270/275/276/28x/29x +====== + laptop-amic Laptops with analog-mic input + laptop-dmic Laptops with digital-mic input + alc269-dmic Enable ALC269(VA) digital mic workaround + alc271-dmic Enable ALC271X digital mic workaround + inv-dmic Inverted internal mic workaround + headset-mic Indicates a combined headset (headphone+mic) jack + lenovo-dock Enables docking station I/O for some Lenovos + dell-headset-multi Headset jack, which can also be used as mic-in + dell-headset-dock Headset jack (without mic-in), and also dock I/O + +ALC66x/67x/892 +============== + mario Chromebook mario model fixup + asus-mode1 ASUS + asus-mode2 ASUS + asus-mode3 ASUS + asus-mode4 ASUS + asus-mode5 ASUS + asus-mode6 ASUS + asus-mode7 ASUS + asus-mode8 ASUS + inv-dmic Inverted internal mic workaround + dell-headset-multi Headset jack, which can also be used as mic-in + +ALC680 +====== + N/A + +ALC88x/898/1150 +====================== + acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G + acer-aspire-8930g Acer Aspire 8330G/6935G + acer-aspire Acer Aspire others + inv-dmic Inverted internal mic workaround + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) + +ALC861/660 +========== + N/A + +ALC861VD/660VD +============== + N/A + +CMI9880 +======= + minimal 3-jack in back + min_fp 3-jack in back, 2-jack in front + full 6-jack in back, 2-jack in front + full_dig 6-jack in back, 2-jack in front, SPDIF I/O + allout 5-jack in back, 2-jack in front, SPDIF out + auto auto-config reading BIOS (default) + +AD1882 / AD1882A +================ + 3stack 3-stack mode + 3stack-automute 3-stack with automute front HP (default) + 6stack 6-stack mode + +AD1884A / AD1883 / AD1984A / AD1984B +==================================== + desktop 3-stack desktop (default) + laptop laptop with HP jack sensing + mobile mobile devices with HP jack sensing + thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart + +AD1884 +====== + N/A + +AD1981 +====== + basic 3-jack (default) + hp HP nx6320 + thinkpad Lenovo Thinkpad T60/X60/Z60 + toshiba Toshiba U205 + +AD1983 +====== + N/A + +AD1984 +====== + basic default configuration + thinkpad Lenovo Thinkpad T61/X61 + dell_desktop Dell T3400 + +AD1986A +======= + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + laptop-imic 2-channel with built-in mic + eapd Turn on EAPD constantly + +AD1988/AD1988B/AD1989A/AD1989B +============================== + 6stack 6-jack + 6stack-dig ditto with SPDIF + 3stack 3-jack + 3stack-dig ditto with SPDIF + laptop 3-jack with hp-jack automute + laptop-dig ditto with SPDIF + auto auto-config reading BIOS (default) + +Conexant 5045 +============= + laptop-hpsense Laptop with HP sense (old model laptop) + laptop-micsense Laptop with Mic sense (old model fujitsu) + laptop-hpmicsense Laptop with HP and Mic senses + benq Benq R55E + laptop-hp530 HP 530 laptop + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + +Conexant 5047 +============= + laptop Basic Laptop config + laptop-hp Laptop config for some HP models (subdevice 30A5) + laptop-eapd Laptop config with EAPD support + test for testing/debugging purpose, almost all controls + can be adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + +Conexant 5051 +============= + laptop Basic Laptop config (default) + hp HP Spartan laptop + hp-dv6736 HP dv6736 + hp-f700 HP Compaq Presario F700 + ideapad Lenovo IdeaPad laptop + toshiba Toshiba Satellite M300 + +Conexant 5066 +============= + laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 + dell-laptop Dell laptops + dell-vostro Dell Vostro + olpc-xo-1_5 OLPC XO 1.5 + ideapad Lenovo IdeaPad U150 + thinkpad Lenovo Thinkpad + +STAC9200 +======== + ref Reference board + oqo OQO Model 2 + dell-d21 Dell (unknown) + dell-d22 Dell (unknown) + dell-d23 Dell (unknown) + dell-m21 Dell Inspiron 630m, Dell Inspiron 640m + dell-m22 Dell Latitude D620, Dell Latitude D820 + dell-m23 Dell XPS M1710, Dell Precision M90 + dell-m24 Dell Latitude 120L + dell-m25 Dell Inspiron E1505n + dell-m26 Dell Inspiron 1501 + dell-m27 Dell Inspiron E1705/9400 + gateway-m4 Gateway laptops with EAPD control + gateway-m4-2 Gateway laptops with EAPD control + panasonic Panasonic CF-74 + auto BIOS setup (default) + +STAC9205/9254 +============= + ref Reference board + dell-m42 Dell (unknown) + dell-m43 Dell Precision + dell-m44 Dell Inspiron + eapd Keep EAPD on (e.g. Gateway T1616) + auto BIOS setup (default) + +STAC9220/9221 +============= + ref Reference board + 3stack D945 3stack + 5stack D945 5stack + SPDIF + intel-mac-v1 Intel Mac Type 1 + intel-mac-v2 Intel Mac Type 2 + intel-mac-v3 Intel Mac Type 3 + intel-mac-v4 Intel Mac Type 4 + intel-mac-v5 Intel Mac Type 5 + intel-mac-auto Intel Mac (detect type according to subsystem id) + macmini Intel Mac Mini (equivalent with type 3) + macbook Intel Mac Book (eq. type 5) + macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) + macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) + imac-intel Intel iMac (eq. type 2) + imac-intel-20 Intel iMac (newer version) (eq. type 3) + ecs202 ECS/PC chips + dell-d81 Dell (unknown) + dell-d82 Dell (unknown) + dell-m81 Dell (unknown) + dell-m82 Dell XPS M1210 + auto BIOS setup (default) + +STAC9202/9250/9251 +================== + ref Reference board, base config + m1 Some Gateway MX series laptops (NX560XL) + m1-2 Some Gateway MX series laptops (MX6453) + m2 Some Gateway MX series laptops (M255) + m2-2 Some Gateway MX series laptops + m3 Some Gateway MX series laptops + m5 Some Gateway MX series laptops (MP6954) + m6 Some Gateway NX series laptops + auto BIOS setup (default) + +STAC9227/9228/9229/927x +======================= + ref Reference board + ref-no-jd Reference board without HP/Mic jack detection + 3stack D965 3stack + 5stack D965 5stack + SPDIF + 5stack-no-fp D965 5stack without front panel + dell-3stack Dell Dimension E520 + dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic + volknob Fixes with volume-knob widget 0x24 + auto BIOS setup (default) + +STAC92HD71B* +============ + ref Reference board + dell-m4-1 Dell desktops + dell-m4-2 Dell desktops + dell-m4-3 Dell desktops + hp-m4 HP mini 1000 + hp-dv5 HP dv series + hp-hdx HP HDX series + hp-dv4-1222nr HP dv4-1222nr (with LED support) + auto BIOS setup (default) + +STAC92HD73* +=========== + ref Reference board + no-jd BIOS setup but without jack-detection + intel Intel DG45* mobos + dell-m6-amic Dell desktops/laptops with analog mics + dell-m6-dmic Dell desktops/laptops with digital mics + dell-m6 Dell desktops/laptops with both type of mics + dell-eq Dell desktops/laptops + alienware Alienware M17x + auto BIOS setup (default) + +STAC92HD83* +=========== + ref Reference board + mic-ref Reference board with power management for ports + dell-s14 Dell laptop + dell-vostro-3500 Dell Vostro 3500 laptop + hp-dv7-4000 HP dv-7 4000 + hp_cNB11_intquad HP CNB models with 4 speakers + hp-zephyr HP Zephyr + hp-led HP with broken BIOS for mute LED + hp-inv-led HP with broken BIOS for inverted mute LED + hp-mic-led HP with mic-mute LED + headset-jack Dell Latitude with a 4-pin headset jack + hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) + hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) + hp-bnb13-eq Hardware equalizer setup for HP laptops + auto BIOS setup (default) + +STAC92HD95 +========== + hp-led LED support for HP laptops + hp-bass Bass HPF setup for HP Spectre 13 + +STAC9872 +======== + vaio VAIO laptop without SPDIF + auto BIOS setup (default) + +Cirrus Logic CS4206/4207 +======================== + mbp55 MacBook Pro 5,5 + imac27 IMac 27 Inch + auto BIOS setup (default) + +Cirrus Logic CS4208 +=================== + mba6 MacBook Air 6,1 and 6,2 + gpio0 Enable GPIO 0 amp + auto BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== + auto BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt new file mode 100644 index 000000000..e7193aac6 --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -0,0 +1,863 @@ +MORE NOTES ON HD-AUDIO DRIVER +============================= + Takashi Iwai <tiwai@suse.de> + + +GENERAL +------- + +HD-audio is the new standard on-board audio component on modern PCs +after AC97. Although Linux has been supporting HD-audio since long +time ago, there are often problems with new machines. A part of the +problem is broken BIOS, and the rest is the driver implementation. +This document explains the brief trouble-shooting and debugging +methods for the HD-audio hardware. + +The HD-audio component consists of two parts: the controller chip and +the codec chips on the HD-audio bus. Linux provides a single driver +for all controllers, snd-hda-intel. Although the driver name contains +a word of a well-known hardware vendor, it's not specific to it but for +all controller chips by other companies. Since the HD-audio +controllers are supposed to be compatible, the single snd-hda-driver +should work in most cases. But, not surprisingly, there are known +bugs and issues specific to each controller type. The snd-hda-intel +driver has a bunch of workarounds for these as described below. + +A controller may have multiple codecs. Usually you have one audio +codec and optionally one modem codec. In theory, there might be +multiple audio codecs, e.g. for analog and digital outputs, and the +driver might not work properly because of conflict of mixer elements. +This should be fixed in future if such hardware really exists. + +The snd-hda-intel driver has several different codec parsers depending +on the codec. It has a generic parser as a fallback, but this +functionality is fairly limited until now. Instead of the generic +parser, usually the codec-specific parser (coded in patch_*.c) is used +for the codec-specific implementations. The details about the +codec-specific problems are explained in the later sections. + +If you are interested in the deep debugging of HD-audio, read the +HD-audio specification at first. The specification is found on +Intel's web page, for example: + +- http://www.intel.com/standards/hdaudio/ + + +HD-AUDIO CONTROLLER +------------------- + +DMA-Position Problem +~~~~~~~~~~~~~~~~~~~~ +The most common problem of the controller is the inaccurate DMA +pointer reporting. The DMA pointer for playback and capture can be +read in two ways, either via a LPIB register or via a position-buffer +map. As default the driver tries to read from the io-mapped +position-buffer, and falls back to LPIB if the position-buffer appears +dead. However, this detection isn't perfect on some devices. In such +a case, you can change the default method via `position_fix` option. + +`position_fix=1` means to use LPIB method explicitly. +`position_fix=2` means to use the position-buffer. +`position_fix=3` means to use a combination of both methods, needed +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +`position_fix=4` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other +controllers, the automatic check and fallback to LPIB as described in +the above. If you get a problem of repeated sounds, this option might +help. + +In addition to that, every controller is known to be broken regarding +the wake-up timing. It wakes up a few samples before actually +processing the data on the buffer. This caused a lot of problems, for +example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts +an artificial delay to the wake up timing. This delay is controlled +via `bdl_pos_adj` option. + +When `bdl_pos_adj` is a negative value (as default), it's assigned to +an appropriate value depending on the controller chip. For Intel +chips, it'd be 1 while it'd be 32 for others. Usually this works. +Only in case it doesn't work and you get warning messages, you should +change this parameter to other values. + + +Codec-Probing Problem +~~~~~~~~~~~~~~~~~~~~~ +A less often but a more severe problem is the codec probing. When +BIOS reports the available codec slots wrongly, the driver gets +confused and tries to access the non-existing codec slot. This often +results in the total screw-up, and destructs the further communication +with the codec chips. The symptom appears usually as error messages +like: +------------------------------------------------------------------------ + hda_intel: azx_get_response timeout, switching to polling mode: + last cmd=0x12345678 + hda_intel: azx_get_response timeout, switching to single_cmd mode: + last cmd=0x12345678 +------------------------------------------------------------------------ + +The first line is a warning, and this is usually relatively harmless. +It means that the codec response isn't notified via an IRQ. The +driver uses explicit polling method to read the response. It gives +very slight CPU overhead, but you'd unlikely notice it. + +The second line is, however, a fatal error. If this happens, usually +it means that something is really wrong. Most likely you are +accessing a non-existing codec slot. + +Thus, if the second error message appears, try to narrow the probed +codec slots via `probe_mask` option. It's a bitmask, and each bit +corresponds to the codec slot. For example, to probe only the first +slot, pass `probe_mask=1`. For the first and the third slots, pass +`probe_mask=5` (where 5 = 1 | 4), and so on. + +Since 2.6.29 kernel, the driver has a more robust probing method, so +this error might happen rarely, though. + +On a machine with a broken BIOS, sometimes you need to force the +driver to probe the codec slots the hardware doesn't report for use. +In such a case, turn the bit 8 (0x100) of `probe_mask` option on. +Then the rest 8 bits are passed as the codec slots to probe +unconditionally. For example, `probe_mask=0x103` will force to probe +the codec slots 0 and 1 no matter what the hardware reports. + + +Interrupt Handling +~~~~~~~~~~~~~~~~~~ +HD-audio driver uses MSI as default (if available) since 2.6.33 +kernel as MSI works better on some machines, and in general, it's +better for performance. However, Nvidia controllers showed bad +regressions with MSI (especially in a combination with AMD chipset), +thus we disabled MSI for them. + +There seem also still other devices that don't work with MSI. If you +see a regression wrt the sound quality (stuttering, etc) or a lock-up +in the recent kernel, try to pass `enable_msi=0` option to disable +MSI. If it works, you can add the known bad device to the blacklist +defined in hda_intel.c. In such a case, please report and give the +patch back to the upstream developer. + + +HD-AUDIO CODEC +-------------- + +Model Option +~~~~~~~~~~~~ +The most common problem regarding the HD-audio driver is the +unsupported codec features or the mismatched device configuration. +Most of codec-specific code has several preset models, either to +override the BIOS setup or to provide more comprehensive features. + +The driver checks PCI SSID and looks through the static configuration +table until any matching entry is found. If you have a new machine, +you may see a message like below: +------------------------------------------------------------------------ + hda_codec: ALC880: BIOS auto-probing. +------------------------------------------------------------------------ +Meanwhile, in the earlier versions, you would see a message like: +------------------------------------------------------------------------ + hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... +------------------------------------------------------------------------ +Even if you see such a message, DON'T PANIC. Take a deep breath and +keep your towel. First of all, it's an informational message, no +warning, no error. This means that the PCI SSID of your device isn't +listed in the known preset model (white-)list. But, this doesn't mean +that the driver is broken. Many codec-drivers provide the automatic +configuration mechanism based on the BIOS setup. + +The HD-audio codec has usually "pin" widgets, and BIOS sets the default +configuration of each pin, which indicates the location, the +connection type, the jack color, etc. The HD-audio driver can guess +the right connection judging from these default configuration values. +However -- some codec-support codes, such as patch_analog.c, don't +support the automatic probing (yet as of 2.6.28). And, BIOS is often, +yes, pretty often broken. It sets up wrong values and screws up the +driver. + +The preset model (or recently called as "fix-up") is provided +basically to overcome such a situation. When the matching preset +model is found in the white-list, the driver assumes the static +configuration of that preset with the correct pin setup, etc. +Thus, if you have a newer machine with a slightly different PCI SSID +(or codec SSID) from the existing one, you may have a good chance to +re-use the same model. You can pass the `model` option to specify the +preset model instead of PCI (and codec-) SSID look-up. + +What `model` option values are available depends on the codec chip. +Check your codec chip from the codec proc file (see "Codec Proc-File" +section below). It will show the vendor/product name of your codec +chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file, +the section of HD-audio driver. You can find a list of codecs +and `model` options belonging to each codec. For example, for Realtek +ALC262 codec chip, pass `model=ultra` for devices that are compatible +with Samsung Q1 Ultra. + +Thus, the first thing you can do for any brand-new, unsupported and +non-working HD-audio hardware is to check HD-audio codec and several +different `model` option values. If you have any luck, some of them +might suit with your device well. + +There are a few special model option values: +- when 'nofixup' is passed, the device-specific fixups in the codec + parser are skipped. +- when `generic` is passed, the codec-specific parser is skipped and + only the generic parser is used. + + +Speaker and Headphone Output +~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +One of the most frequent (and obvious) bugs with HD-audio is the +silent output from either or both of a built-in speaker and a +headphone jack. In general, you should try a headphone output at +first. A speaker output often requires more additional controls like +the external amplifier bits. Thus a headphone output has a slightly +better chance. + +Before making a bug report, double-check whether the mixer is set up +correctly. The recent version of snd-hda-intel driver provides mostly +"Master" volume control as well as "Front" volume (where Front +indicates the front-channels). In addition, there can be individual +"Headphone" and "Speaker" controls. + +Ditto for the speaker output. There can be "External Amplifier" +switch on some codecs. Turn on this if present. + +Another related problem is the automatic mute of speaker output by +headphone plugging. This feature is implemented in most cases, but +not on every preset model or codec-support code. + +In anyway, try a different model option if you have such a problem. +Some other models may match better and give you more matching +functionality. If none of the available models works, send a bug +report. See the bug report section for details. + +If you are masochistic enough to debug the driver problem, note the +following: + +- The speaker (and the headphone, too) output often requires the + external amplifier. This can be set usually via EAPD verb or a + certain GPIO. If the codec pin supports EAPD, you have a better + chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly + it's either GPIO0 or GPIO1) may turn on/off EAPD. +- Some Realtek codecs require special vendor-specific coefficients to + turn on the amplifier. See patch_realtek.c. +- IDT codecs may have extra power-enable/disable controls on each + analog pin. See patch_sigmatel.c. +- Very rare but some devices don't accept the pin-detection verb until + triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the + codec-communication stall. Some examples are found in + patch_realtek.c. + + +Capture Problems +~~~~~~~~~~~~~~~~ +The capture problems are often because of missing setups of mixers. +Thus, before submitting a bug report, make sure that you set up the +mixer correctly. For example, both "Capture Volume" and "Capture +Switch" have to be set properly in addition to the right "Capture +Source" or "Input Source" selection. Some devices have "Mic Boost" +volume or switch. + +When the PCM device is opened via "default" PCM (without pulse-audio +plugin), you'll likely have "Digital Capture Volume" control as well. +This is provided for the extra gain/attenuation of the signal in +software, especially for the inputs without the hardware volume +control such as digital microphones. Unless really needed, this +should be set to exactly 50%, corresponding to 0dB -- neither extra +gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM, +this control will have no influence, though. + +It's known that some codecs / devices have fairly bad analog circuits, +and the recorded sound contains a certain DC-offset. This is no bug +of the driver. + +Most of modern laptops have no analog CD-input connection. Thus, the +recording from CD input won't work in many cases although the driver +provides it as the capture source. Use CDDA instead. + +The automatic switching of the built-in and external mic per plugging +is implemented on some codec models but not on every model. Partly +because of my laziness but mostly lack of testers. Feel free to +submit the improvement patch to the author. + + +Direct Debugging +~~~~~~~~~~~~~~~~ +If no model option gives you a better result, and you are a tough guy +to fight against evil, try debugging via hitting the raw HD-audio +codec verbs to the device. Some tools are available: hda-emu and +hda-analyzer. The detailed description is found in the sections +below. You'd need to enable hwdep for using these tools. See "Kernel +Configuration" section. + + +OTHER ISSUES +------------ + +Kernel Configuration +~~~~~~~~~~~~~~~~~~~~ +In general, I recommend you to enable the sound debug option, +`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not. +This enables snd_printd() macro and others, and you'll get additional +kernel messages at probing. + +In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this +will give you far more messages. Thus turn this on only when you are +sure to want it. + +Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*` +options. Note that each of them corresponds to the codec chip, not +the controller chip. Thus, even if lspci shows the Nvidia controller, +you may need to choose the option for other vendors. If you are +unsure, just select all yes. + +`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver. +When this is enabled, the driver creates hardware-dependent devices +(one per each codec), and you have a raw access to the device via +these device files. For example, `hwC0D2` will be created for the +codec slot #2 of the first card (#0). For debug-tools such as +hda-verb and hda-analyzer, the hwdep device has to be enabled. +Thus, it'd be better to turn this on always. + +`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the +hwdep option above. When enabled, you'll have some sysfs files under +the corresponding hwdep directory. See "HD-audio reconfiguration" +section below. + +`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature. +See "Power-saving" section below. + + +Codec Proc-File +~~~~~~~~~~~~~~~ +The codec proc-file is a treasure-chest for debugging HD-audio. +It shows most of useful information of each codec widget. + +The proc file is located in /proc/asound/card*/codec#*, one file per +each codec slot. You can know the codec vendor, product id and +names, the type of each widget, capabilities and so on. +This file, however, doesn't show the jack sensing state, so far. This +is because the jack-sensing might be depending on the trigger state. + +This file will be picked up by the debug tools, and also it can be fed +to the emulator as the primary codec information. See the debug tools +section below. + +This proc file can be also used to check whether the generic parser is +used. When the generic parser is used, the vendor/product ID name +will appear as "Realtek ID 0262", instead of "Realtek ALC262". + + +HD-Audio Reconfiguration +~~~~~~~~~~~~~~~~~~~~~~~~ +This is an experimental feature to allow you re-configure the HD-audio +codec dynamically without reloading the driver. The following sysfs +files are available under each codec-hwdep device directory (e.g. +/sys/class/sound/hwC0D0): + +vendor_id:: + Shows the 32bit codec vendor-id hex number. You can change the + vendor-id value by writing to this file. +subsystem_id:: + Shows the 32bit codec subsystem-id hex number. You can change the + subsystem-id value by writing to this file. +revision_id:: + Shows the 32bit codec revision-id hex number. You can change the + revision-id value by writing to this file. +afg:: + Shows the AFG ID. This is read-only. +mfg:: + Shows the MFG ID. This is read-only. +name:: + Shows the codec name string. Can be changed by writing to this + file. +modelname:: + Shows the currently set `model` option. Can be changed by writing + to this file. +init_verbs:: + The extra verbs to execute at initialization. You can add a verb by + writing to this file. Pass three numbers: nid, verb and parameter + (separated with a space). +hints:: + Shows / stores hint strings for codec parsers for any use. + Its format is `key = value`. For example, passing `jack_detect = no` + will disable the jack detection of the machine completely. +init_pin_configs:: + Shows the initial pin default config values set by BIOS. +driver_pin_configs:: + Shows the pin default values set by the codec parser explicitly. + This doesn't show all pin values but only the changed values by + the parser. That is, if the parser doesn't change the pin default + config values by itself, this will contain nothing. +user_pin_configs:: + Shows the pin default config values to override the BIOS setup. + Writing this (with two numbers, NID and value) appends the new + value. The given will be used instead of the initial BIOS value at + the next reconfiguration time. Note that this config will override + even the driver pin configs, too. +reconfig:: + Triggers the codec re-configuration. When any value is written to + this file, the driver re-initialize and parses the codec tree + again. All the changes done by the sysfs entries above are taken + into account. +clear:: + Resets the codec, removes the mixer elements and PCM stuff of the + specified codec, and clear all init verbs and hints. + +For example, when you want to change the pin default configuration +value of the pin widget 0x14 to 0x9993013f, and let the driver +re-configure based on that state, run like below: +------------------------------------------------------------------------ + # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs + # echo 1 > /sys/class/sound/hwC0D0/reconfig +------------------------------------------------------------------------ + + +Hint Strings +~~~~~~~~~~~~ +The codec parser have several switches and adjustment knobs for +matching better with the actual codec or device behavior. Many of +them can be adjusted dynamically via "hints" strings as mentioned in +the section above. For example, by passing `jack_detect = no` string +via sysfs or a patch file, you can disable the jack detection, thus +the codec parser will skip the features like auto-mute or mic +auto-switch. As a boolean value, either `yes`, `no`, `true`, `false`, +`1` or `0` can be passed. + +The generic parser supports the following hints: + +- jack_detect (bool): specify whether the jack detection is available + at all on this machine; default true +- inv_jack_detect (bool): indicates that the jack detection logic is + inverted +- trigger_sense (bool): indicates that the jack detection needs the + explicit call of AC_VERB_SET_PIN_SENSE verb +- inv_eapd (bool): indicates that the EAPD is implemented in the + inverted logic +- pcm_format_first (bool): sets the PCM format before the stream tag + and channel ID +- sticky_stream (bool): keep the PCM format, stream tag and ID as long + as possible; default true +- spdif_status_reset (bool): reset the SPDIF status bits at each time + the SPDIF stream is set up +- pin_amp_workaround (bool): the output pin may have multiple amp + values +- single_adc_amp (bool): ADCs can have only single input amps +- auto_mute (bool): enable/disable the headphone auto-mute feature; + default true +- auto_mic (bool): enable/disable the mic auto-switch feature; default + true +- line_in_auto_switch (bool): enable/disable the line-in auto-switch + feature; default false +- need_dac_fix (bool): limits the DACs depending on the channel count +- primary_hp (bool): probe headphone jacks as the primary outputs; + default true +- multi_io (bool): try probing multi-I/O config (e.g. shared + line-in/surround, mic/clfe jacks) +- multi_cap_vol (bool): provide multiple capture volumes +- inv_dmic_split (bool): provide split internal mic volume/switch for + phase-inverted digital mics +- indep_hp (bool): provide the independent headphone PCM stream and + the corresponding mixer control, if available +- add_stereo_mix_input (bool): add the stereo mix (analog-loopback + mix) to the input mux if available +- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each + I/O jack for allowing to change the headphone amp and mic bias VREF + capabilities +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future +- add_hp_mic (bool): add the headphone to capture source if possible +- hp_mic_detect (bool): enable/disable the hp/mic shared input for a + single built-in mic case; default true +- mixer_nid (int): specifies the widget NID of the analog-loopback + mixer + + +Early Patching +~~~~~~~~~~~~~~ +When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a +firmware file for modifying the HD-audio setup before initializing the +codec. This can work basically like the reconfiguration via sysfs in +the above, but it does it before the first codec configuration. + +A patch file is a plain text file which looks like below: + +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [model] + auto + + [pincfg] + 0x12 0x411111f0 + + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff + + [hint] + jack_detect = no +------------------------------------------------------------------------ + +The file needs to have a line `[codec]`. The next line should contain +three numbers indicating the codec vendor-id (0x12345678 in the +example), the codec subsystem-id (0xabcd1234) and the address (2) of +the codec. The rest patch entries are applied to this specified codec +until another codec entry is given. Passing 0 or a negative number to +the first or the second value will make the check of the corresponding +field be skipped. It'll be useful for really broken devices that don't +initialize SSID properly. + +The `[model]` line allows to change the model name of the each codec. +In the example above, it will be changed to model=auto. +Note that this overrides the module option. + +After the `[pincfg]` line, the contents are parsed as the initial +default pin-configurations just like `user_pin_configs` sysfs above. +The values can be shown in user_pin_configs sysfs file, too. + +Similarly, the lines after `[verb]` are parsed as `init_verbs` +sysfs entries, and the lines after `[hint]` are parsed as `hints` +sysfs entries, respectively. + +Another example to override the codec vendor id from 0x12345678 to +0xdeadbeef is like below: +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [vendor_id] + 0xdeadbeef +------------------------------------------------------------------------ + +In the similar way, you can override the codec subsystem_id via +`[subsystem_id]`, the revision id via `[revision_id]` line. +Also, the codec chip name can be rewritten via `[chip_name]` line. +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [subsystem_id] + 0xffff1111 + + [revision_id] + 0x10 + + [chip_name] + My-own NEWS-0002 +------------------------------------------------------------------------ + +The hd-audio driver reads the file via request_firmware(). Thus, +a patch file has to be located on the appropriate firmware path, +typically, /lib/firmware. For example, when you pass the option +`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be +present. + +The patch module option is specific to each card instance, and you +need to give one file name for each instance, separated by commas. +For example, if you have two cards, one for an on-board analog and one +for an HDMI video board, you may pass patch option like below: +------------------------------------------------------------------------ + options snd-hda-intel patch=on-board-patch,hdmi-patch +------------------------------------------------------------------------ + + +Power-Saving +~~~~~~~~~~~~ +The power-saving is a kind of auto-suspend of the device. When the +device is inactive for a certain time, the device is automatically +turned off to save the power. The time to go down is specified via +`power_save` module option, and this option can be changed dynamically +via sysfs. + +The power-saving won't work when the analog loopback is enabled on +some codecs. Make sure that you mute all unneeded signal routes when +you want the power-saving. + +The power-saving feature might cause audible click noises at each +power-down/up depending on the device. Some of them might be +solvable, but some are hard, I'm afraid. Some distros such as +openSUSE enables the power-saving feature automatically when the power +cable is unplugged. Thus, if you hear noises, suspect first the +power-saving. See /sys/module/snd_hda_intel/parameters/power_save to +check the current value. If it's non-zero, the feature is turned on. + +The recent kernel supports the runtime PM for the HD-audio controller +chip, too. It means that the HD-audio controller is also powered up / +down dynamically. The feature is enabled only for certain controller +chips like Intel LynxPoint. You can enable/disable this feature +forcibly by setting `power_save_controller` option, which is also +available at /sys/module/snd_hda_intel/parameters directory. + + +Tracepoints +~~~~~~~~~~~ +The hd-audio driver gives a few basic tracepoints. +`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response` +traces the response from RIRB (only when read from the codec driver). +`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc, +`hda:hda_unsol_event` traces the unsolicited events, and +`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up +via power-saving behavior. + +Enabling all tracepoints can be done like +------------------------------------------------------------------------ + # echo 1 > /sys/kernel/debug/tracing/events/hda/enable +------------------------------------------------------------------------ +then after some commands, you can traces from +/sys/kernel/debug/tracing/trace file. For example, when you want to +trace what codec command is sent, enable the tracepoint like: +------------------------------------------------------------------------ + # cat /sys/kernel/debug/tracing/trace + # tracer: nop + # + # TASK-PID CPU# TIMESTAMP FUNCTION + # | | | | | + <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019 + <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019 + <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a + <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a + <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019 + <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019 + <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a + <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a +------------------------------------------------------------------------ +Here `[0:0]` indicates the card number and the codec address, and +`val` shows the value sent to the codec, respectively. The value is +a packed value, and you can decode it via hda-decode-verb program +included in hda-emu package below. For example, the value e3a019 is +to set the left output-amp value to 25. +------------------------------------------------------------------------ + % hda-decode-verb 0xe3a019 + raw value = 0x00e3a019 + cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19 + raw value: verb = 0x3a0, parm = 0x19 + verbname = set_amp_gain_mute + amp raw val = 0xa019 + output, left, idx=0, mute=0, val=25 +------------------------------------------------------------------------ + + +Development Tree +~~~~~~~~~~~~~~~~ +The latest development codes for HD-audio are found on sound git tree: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git + +The master branch or for-next branches can be used as the main +development branches in general while the development for the current +and next kernels are found in for-linus and for-next branches, +respectively. + +If you are using the latest Linus tree, it'd be better to pull the +above GIT tree onto it. If you are using the older kernels, an easy +way to try the latest ALSA code is to build from the snapshot +tarball. There are daily tarballs and the latest snapshot tarball. +All can be built just like normal alsa-driver release packages, that +is, installed via the usual spells: configure, make and make +install(-modules). See INSTALL in the package. The snapshot tarballs +are found at: + +- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ + + +Sending a Bug Report +~~~~~~~~~~~~~~~~~~~~ +If any model or module options don't work for your device, it's time +to send a bug report to the developers. Give the following in your +bug report: + +- Hardware vendor, product and model names +- Kernel version (and ALSA-driver version if you built externally) +- `alsa-info.sh` output; run with `--no-upload` option. See the + section below about alsa-info + +If it's a regression, at best, send alsa-info outputs of both working +and non-working kernels. This is really helpful because we can +compare the codec registers directly. + +Send a bug report either the followings: + +kernel-bugzilla:: + https://bugzilla.kernel.org/ +alsa-devel ML:: + alsa-devel@alsa-project.org + + +DEBUG TOOLS +----------- + +This section describes some tools available for debugging HD-audio +problems. + +alsa-info +~~~~~~~~~ +The script `alsa-info.sh` is a very useful tool to gather the audio +device information. You can fetch the latest version from: + +- http://www.alsa-project.org/alsa-info.sh + +Run this script as root, and it will gather the important information +such as the module lists, module parameters, proc file contents +including the codec proc files, mixer outputs and the control +elements. As default, it will store the information onto a web server +on alsa-project.org. But, if you send a bug report, it'd be better to +run with `--no-upload` option, and attach the generated file. + +There are some other useful options. See `--help` option output for +details. + +When a probe error occurs or when the driver obviously assigns a +mismatched model, it'd be helpful to load the driver with +`probe_only=1` option (at best after the cold reboot) and run +alsa-info at this state. With this option, the driver won't configure +the mixer and PCM but just tries to probe the codec slot. After +probing, the proc file is available, so you can get the raw codec +information before modified by the driver. Of course, the driver +isn't usable with `probe_only=1`. But you can continue the +configuration via hwdep sysfs file if hda-reconfig option is enabled. +Using `probe_only` mask 2 skips the reset of HDA codecs (use +`probe_only=3` as module option). The hwdep interface can be used +to determine the BIOS codec initialization. + + +hda-verb +~~~~~~~~ +hda-verb is a tiny program that allows you to access the HD-audio +codec directly. You can execute a raw HD-audio codec verb with this. +This program accesses the hwdep device, thus you need to enable the +kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand. + +The hda-verb program takes four arguments: the hwdep device file, the +widget NID, the verb and the parameter. When you access to the codec +on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first +argument, typically. (However, the real path name depends on the +system.) + +The second parameter is the widget number-id to access. The third +parameter can be either a hex/digit number or a string corresponding +to a verb. Similarly, the last parameter is the value to write, or +can be a string for the parameter type. + +------------------------------------------------------------------------ + % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 + nid = 0x12, verb = 0x701, param = 0x2 + value = 0x0 + + % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID + nid = 0x0, verb = 0xf00, param = 0x0 + value = 0x10ec0262 + + % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 + nid = 0x2, verb = 0x300, param = 0xb080 + value = 0x0 +------------------------------------------------------------------------ + +Although you can issue any verbs with this program, the driver state +won't be always updated. For example, the volume values are usually +cached in the driver, and thus changing the widget amp value directly +via hda-verb won't change the mixer value. + +The hda-verb program is included now in alsa-tools: + +- git://git.alsa-project.org/alsa-tools.git + +Also, the old stand-alone package is found in the ftp directory: + +- ftp://ftp.suse.com/pub/people/tiwai/misc/ + +Also a git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git + +See README file in the tarball for more details about hda-verb +program. + + +hda-analyzer +~~~~~~~~~~~~ +hda-analyzer provides a graphical interface to access the raw HD-audio +control, based on pyGTK2 binding. It's a more powerful version of +hda-verb. The program gives you an easy-to-use GUI stuff for showing +the widget information and adjusting the amp values, as well as the +proc-compatible output. + +The hda-analyzer: + +- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer + +is a part of alsa.git repository in alsa-project.org: + +- git://git.alsa-project.org/alsa.git + +Codecgraph +~~~~~~~~~~ +Codecgraph is a utility program to generate a graph and visualizes the +codec-node connection of a codec chip. It's especially useful when +you analyze or debug a codec without a proper datasheet. The program +parses the given codec proc file and converts to SVG via graphiz +program. + +The tarball and GIT trees are found in the web page at: + +- http://helllabs.org/codecgraph/ + + +hda-emu +~~~~~~~ +hda-emu is an HD-audio emulator. The main purpose of this program is +to debug an HD-audio codec without the real hardware. Thus, it +doesn't emulate the behavior with the real audio I/O, but it just +dumps the codec register changes and the ALSA-driver internal changes +at probing and operating the HD-audio driver. + +The program requires a codec proc-file to simulate. Get a proc file +for the target codec beforehand, or pick up an example codec from the +codec proc collections in the tarball. Then, run the program with the +proc file, and the hda-emu program will start parsing the codec file +and simulates the HD-audio driver: + +------------------------------------------------------------------------ + % hda-emu codecs/stac9200-dell-d820-laptop + # Parsing.. + hda_codec: Unknown model for STAC9200, using BIOS defaults + hda_codec: pin nid 08 bios pin config 40c003fa + .... +------------------------------------------------------------------------ + +The program gives you only a very dumb command-line interface. You +can get a proc-file dump at the current state, get a list of control +(mixer) elements, set/get the control element value, simulate the PCM +operation, the jack plugging simulation, etc. + +The package is found in: + +- ftp://ftp.suse.com/pub/people/tiwai/misc/ + +A git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git + +See README file in the tarball for more details about hda-emu +program. + + +hda-jack-retask +~~~~~~~~~~~~~~~ +hda-jack-retask is a user-friendly GUI program to manipulate the +HD-audio pin control for jack retasking. If you have a problem about +the jack assignment, try this program and check whether you can get +useful results. Once when you figure out the proper pin assignment, +it can be fixed either in the driver code statically or via passing a +firmware patch file (see "Early Patching" section). + +The program is included in alsa-tools now: + +- git://git.alsa-project.org/alsa-tools.git + diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt new file mode 100644 index 000000000..ccda41b10 --- /dev/null +++ b/Documentation/sound/alsa/Joystick.txt @@ -0,0 +1,86 @@ +Analog Joystick Support on ALSA Drivers +======================================= + Oct. 14, 2003 + Takashi Iwai <tiwai@suse.de> + +General +------- + +First of all, you need to enable GAMEPORT support on Linux kernel for +using a joystick with the ALSA driver. For the details of gameport +support, refer to Documentation/input/joystick.txt. + +The joystick support of ALSA drivers is different between ISA and PCI +cards. In the case of ISA (PnP) cards, it's usually handled by the +independent module (ns558). Meanwhile, the ALSA PCI drivers have the +built-in gameport support. Hence, when the ALSA PCI driver is built +in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the +gameport support on that card will be (silently) disabled. + +Some adapter modules probe the physical connection of the device at +the load time. It'd be safer to plug in the joystick device before +loading the module. + + +PCI Cards +--------- + +For PCI cards, the joystick is enabled when the appropriate module +option is specified. Some drivers don't need options, and the +joystick support is always enabled. In the former ALSA version, there +was a dynamic control API for the joystick activation. It was +changed, however, to the static module options because of the system +stability and the resource management. + +The following PCI drivers support the joystick natively. + + Driver Module Option Available Values + --------------------------------------------------------------------------- + als4000 joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + au88x0 N/A N/A + azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) + ens1370 joystick 0 = disable (default), 1 = enable + ens1371 joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x200, 0x208, 0x210, 0x218 + cmipci joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + cs4281 N/A N/A + cs46xx N/A N/A + es1938 N/A N/A + es1968 joystick 0 = disable (default), 1 = enable + sonicvibes N/A N/A + trident N/A N/A + via82xx(*1) joystick 0 = disable (default), 1 = enable + ymfpci joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x201, 0x202, 0x204, 0x205(*2) + --------------------------------------------------------------------------- + + *1) VIA686A/B only + *2) With YMF744/754 chips, the port address can be chosen arbitrarily + +The following drivers don't support gameport natively, but there are +additional modules. Load the corresponding module to add the gameport +support. + + Driver Additional Module + ----------------------------- + emu10k1 emu10k1-gp + fm801 fm801-gp + ----------------------------- + +Note: the "pcigame" and "cs461x" modules are for the OSS drivers only. + These ALSA drivers (cs46xx, trident and au88x0) have the + built-in gameport support. + +As mentioned above, ALSA PCI drivers have the built-in gameport +support, so you don't have to load ns558 module. Just load "joydev" +and the appropriate adapter module (e.g. "analog"). + + +ISA Cards +--------- + +ALSA ISA drivers don't have the built-in gameport support. +Instead, you need to load "ns558" module in addition to "joydev" and +the adapter module (e.g. "analog"). diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt new file mode 100644 index 000000000..4ee35b4fb --- /dev/null +++ b/Documentation/sound/alsa/MIXART.txt @@ -0,0 +1,100 @@ + Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards + Digigram <alsa@digigram.com> + + +GENERAL +======= + +The miXart8 is a multichannel audio processing and mixing soundcard +that has 4 stereo audio inputs and 4 stereo audio outputs. +The miXart8AES/EBU is the same with a add-on card that offers further +4 digital stereo audio inputs and outputs. +Furthermore the add-on card offers external clock synchronisation +(AES/EBU, Word Clock, Time Code and Video Synchro) + +The mainboard has a PowerPC that offers onboard mpeg encoding and +decoding, samplerate conversions and various effects. + +The driver don't work properly at all until the certain firmwares +are loaded, i.e. no PCM nor mixer devices will appear. +Use the mixartloader that can be found in the alsa-tools package. + + +VERSION 0.1.0 +============= + +One miXart8 board will be represented as 4 alsa cards, each with 1 +stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device. +With a miXart8AES/EBU there is in addition 1 stereo digital input +'pcm1c' and 1 stereo digital output 'pcm1p' per card. + +Formats +------- +U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE +Sample rates : 8000 - 48000 Hz continuously + +Playback +-------- +For instance the playback devices are configured to have max. 4 +substreams performing hardware mixing. This could be changed to a +maximum of 24 substreams if wished. +Mono files will be played on the left and right channel. Each channel +can be muted for each stream to use 8 analog/digital outputs separately. + +Capture +------- +There is one substream per capture device. For instance only stereo +formats are supported. + +Mixer +----- +<Master> and <Master Capture> : analog volume control of playback and capture PCM. +<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream. +<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream. +<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume +and mute control. + +Rem : for best audio quality try to keep a 0 attenuation on the PCM +and AES volume controls which is set by 219 in the range from 0 to 255 +(about 86% with alsamixer) + + +NOT YET IMPLEMENTED +=================== + +- external clock support (AES/EBU, Word Clock, Time Code, Video Sync) +- MPEG audio formats +- mono record +- on-board effects and samplerate conversions +- linked streams + + +FIRMWARE +======== + +[As of 2.6.11, the firmware can be loaded automatically with hotplug + when CONFIG_FW_LOADER is set. The mixartloader is necessary only + for older versions or when you build the driver into kernel.] + +For loading the firmware automatically after the module is loaded, use a +install command. For example, add the following entry to +/etc/modprobe.d/mixart.conf for miXart driver: + + install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ + /usr/bin/mixartloader +(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to + /etc/modules.conf, instead.) + +The firmware binaries are installed on /usr/share/alsa/firmware +(or /usr/local/share/alsa/firmware, depending to the prefix option of +configure). There will be a miXart.conf file, which define the dsp image +files. + +The firmware files are copyright by Digigram SA + + +COPYRIGHT +========= + +Copyright (c) 2003 Digigram SA <alsa@digigram.com> +Distributable under GPL. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt new file mode 100644 index 000000000..152ca2a3f --- /dev/null +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -0,0 +1,305 @@ + NOTES ON KERNEL OSS-EMULATION + ============================= + + Jan. 22, 2004 Takashi Iwai <tiwai@suse.de> + + +Modules +======= + +ALSA provides a powerful OSS emulation on the kernel. +The OSS emulation for PCM, mixer and sequencer devices is implemented +as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. +When you need to access the OSS PCM, mixer or sequencer devices, the +corresponding module has to be loaded. + +These modules are loaded automatically when the corresponding service +is called. The alias is defined sound-service-x-y, where x and y are +the card number and the minor unit number. Usually you don't have to +define these aliases by yourself. + +Only necessary step for auto-loading of OSS modules is to define the +card alias in /etc/modprobe.d/alsa.conf, such as + + alias sound-slot-0 snd-emu10k1 + +As the second card, define sound-slot-1 as well. +Note that you can't use the aliased name as the target name (i.e. +"alias sound-slot-0 snd-card-0" doesn't work any more like the old +modutils). + +The currently available OSS configuration is shown in +/proc/asound/oss/sndstat. This shows in the same syntax of +/dev/sndstat, which is available on the commercial OSS driver. +On ALSA, you can symlink /dev/sndstat to this proc file. + +Please note that the devices listed in this proc file appear only +after the corresponding OSS-emulation module is loaded. Don't worry +even if "NOT ENABLED IN CONFIG" is shown in it. + + +Device Mapping +============== + +ALSA supports the following OSS device files: + + PCM: + /dev/dspX + /dev/adspX + + Mixer: + /dev/mixerX + + MIDI: + /dev/midi0X + /dev/amidi0X + + Sequencer: + /dev/sequencer + /dev/sequencer2 (aka /dev/music) + +where X is the card number from 0 to 7. + +(NOTE: Some distributions have the device files like /dev/midi0 and + /dev/midi1. They are NOT for OSS but for tclmidi, which is + a totally different thing.) + +Unlike the real OSS, ALSA cannot use the device files more than the +assigned ones. For example, the first card cannot use /dev/dsp1 or +/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. + +As seen above, PCM and MIDI may have two devices. Usually, the first +PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary +device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and +/dev/amidi, respectively. + +You can change this device mapping via the module options of +snd-pcm-oss and snd-rawmidi. In the case of PCM, the following +options are available for snd-pcm-oss: + + dsp_map PCM device number assigned to /dev/dspX + (default = 0) + adsp_map PCM device number assigned to /dev/adspX + (default = 1) + +For example, to map the third PCM device (hw:0,2) to /dev/adsp0, +define like this: + + options snd-pcm-oss adsp_map=2 + +The options take arrays. For configuring the second card, specify +two entries separated by comma. For example, to map the third PCM +device on the second card to /dev/adsp1, define like below: + + options snd-pcm-oss adsp_map=0,2 + +To change the mapping of MIDI devices, the following options are +available for snd-rawmidi: + + midi_map MIDI device number assigned to /dev/midi0X + (default = 0) + amidi_map MIDI device number assigned to /dev/amidi0X + (default = 1) + +For example, to assign the third MIDI device on the first card to +/dev/midi00, define as follows: + + options snd-rawmidi midi_map=2 + + +PCM Mode +======== + +As default, ALSA emulates the OSS PCM with so-called plugin layer, +i.e. tries to convert the sample format, rate or channels +automatically when the card doesn't support it natively. +This will lead to some problems for some applications like quake or +wine, especially if they use the card only in the MMAP mode. + +In such a case, you can change the behavior of PCM per application by +writing a command to the proc file. There is a proc file for each PCM +stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number +(zero-based), Y the PCM device number (zero-based), and 'p' is for +playback and 'c' for capture, respectively. Note that this proc file +exists only after snd-pcm-oss module is loaded. + +The command sequence has the following syntax: + + app_name fragments fragment_size [options] + +app_name is the name of application with (higher priority) or without +path. +fragments specifies the number of fragments or zero if no specific +number is given. +fragment_size is the size of fragment in bytes or zero if not given. +options is the optional parameters. The following options are +available: + + disable the application tries to open a pcm device for + this channel but does not want to use it. + direct don't use plugins + block force block open mode + non-block force non-block open mode + partial-frag write also partial fragments (affects playback only) + no-silence do not fill silence ahead to avoid clicks + +The disable option is useful when one stream direction (playback or +capture) is not handled correctly by the application although the +hardware itself does support both directions. +The direct option is used, as mentioned above, to bypass the automatic +conversion and useful for MMAP-applications. +For example, to playback the first PCM device without plugins for +quake, send a command via echo like the following: + + % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss + +While quake wants only playback, you may append the second command +to notify driver that only this direction is about to be allocated: + + % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss + +The permission of proc files depend on the module options of snd. +As default it's set as root, so you'll likely need to be superuser for +sending the command above. + +The block and non-block options are used to change the behavior of +opening the device file. + +As default, ALSA behaves as original OSS drivers, i.e. does not block +the file when it's busy. The -EBUSY error is returned in this case. + +This blocking behavior can be changed globally via nonblock_open +module option of snd-pcm-oss. For using the blocking mode as default +for OSS devices, define like the following: + + options snd-pcm-oss nonblock_open=0 + +The partial-frag and no-silence commands have been added recently. +Both commands are for optimization use only. The former command +specifies to invoke the write transfer only when the whole fragment is +filled. The latter stops writing the silence data ahead +automatically. Both are disabled as default. + +You can check the currently defined configuration by reading the proc +file. The read image can be sent to the proc file again, hence you +can save the current configuration + + % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg + +and restore it like + + % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss + +Also, for clearing all the current configuration, send "erase" command +as below: + + % echo "erase" > /proc/asound/card0/pcm0p/oss + + +Mixer Elements +============== + +Since ALSA has completely different mixer interface, the emulation of +OSS mixer is relatively complicated. ALSA builds up a mixer element +from several different ALSA (mixer) controls based on the name +string. For example, the volume element SOUND_MIXER_PCM is composed +from "PCM Playback Volume" and "PCM Playback Switch" controls for the +playback direction and from "PCM Capture Volume" and "PCM Capture +Switch" for the capture directory (if exists). When the PCM volume of +OSS is changed, all the volume and switch controls above are adjusted +automatically. + +As default, ALSA uses the following control for OSS volumes: + + OSS volume ALSA control Index + ----------------------------------------------------- + SOUND_MIXER_VOLUME Master 0 + SOUND_MIXER_BASS Tone Control - Bass 0 + SOUND_MIXER_TREBLE Tone Control - Treble 0 + SOUND_MIXER_SYNTH Synth 0 + SOUND_MIXER_PCM PCM 0 + SOUND_MIXER_SPEAKER PC Speaker 0 + SOUND_MIXER_LINE Line 0 + SOUND_MIXER_MIC Mic 0 + SOUND_MIXER_CD CD 0 + SOUND_MIXER_IMIX Monitor Mix 0 + SOUND_MIXER_ALTPCM PCM 1 + SOUND_MIXER_RECLEV (not assigned) + SOUND_MIXER_IGAIN Capture 0 + SOUND_MIXER_OGAIN Playback 0 + SOUND_MIXER_LINE1 Aux 0 + SOUND_MIXER_LINE2 Aux 1 + SOUND_MIXER_LINE3 Aux 2 + SOUND_MIXER_DIGITAL1 Digital 0 + SOUND_MIXER_DIGITAL2 Digital 1 + SOUND_MIXER_DIGITAL3 Digital 2 + SOUND_MIXER_PHONEIN Phone 0 + SOUND_MIXER_PHONEOUT Phone 1 + SOUND_MIXER_VIDEO Video 0 + SOUND_MIXER_RADIO Radio 0 + SOUND_MIXER_MONITOR Monitor 0 + +The second column is the base-string of the corresponding ALSA +control. In fact, the controls with "XXX [Playback|Capture] +[Volume|Switch]" will be checked in addition. + +The current assignment of these mixer elements is listed in the proc +file, /proc/asound/cardX/oss_mixer, which will be like the following + + VOLUME "Master" 0 + BASS "" 0 + TREBLE "" 0 + SYNTH "" 0 + PCM "PCM" 0 + ... + +where the first column is the OSS volume element, the second column +the base-string of the corresponding ALSA control, and the third the +control index. When the string is empty, it means that the +corresponding OSS control is not available. + +For changing the assignment, you can write the configuration to this +proc file. For example, to map "Wave Playback" to the PCM volume, +send the command like the following: + + % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer + +The command is exactly as same as listed in the proc file. You can +change one or more elements, one volume per line. In the last +example, both "Wave Playback Volume" and "Wave Playback Switch" will +be affected when PCM volume is changed. + +Like the case of PCM proc file, the permission of proc files depend on +the module options of snd. you'll likely need to be superuser for +sending the command above. + +As well as in the case of PCM proc file, you can save and restore the +current mixer configuration by reading and writing the whole file +image. + + +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + +Unsupported Features +==================== + +MMAP on ICE1712 driver +---------------------- +ICE1712 supports only the unconventional format, interleaved +10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap +the buffer as the conventional (mono or 2-channels, 8 or 16bit) format +on OSS. + diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt new file mode 100644 index 000000000..7f8a0d325 --- /dev/null +++ b/Documentation/sound/alsa/Procfile.txt @@ -0,0 +1,234 @@ + Proc Files of ALSA Drivers + ========================== + Takashi Iwai <tiwai@suse.de> + +General +------- + +ALSA has its own proc tree, /proc/asound. Many useful information are +found in this tree. When you encounter a problem and need debugging, +check the files listed in the following sections. + +Each card has its subtree cardX, where X is from 0 to 7. The +card-specific files are stored in the card* subdirectories. + + +Global Information +------------------ + +cards + Shows the list of currently configured ALSA drivers, + index, the id string, short and long descriptions. + +version + Shows the version string and compile date. + +modules + Lists the module of each card + +devices + Lists the ALSA native device mappings. + +meminfo + Shows the status of allocated pages via ALSA drivers. + Appears only when CONFIG_SND_DEBUG=y. + +hwdep + Lists the currently available hwdep devices in format of + <card>-<device>: <name> + +pcm + Lists the currently available PCM devices in format of + <card>-<device>: <id>: <name> : <sub-streams> + +timer + Lists the currently available timer devices + + +oss/devices + Lists the OSS device mappings. + +oss/sndstat + Provides the output compatible with /dev/sndstat. + You can symlink this to /dev/sndstat. + + +Card Specific Files +------------------- + +The card-specific files are found in /proc/asound/card* directories. +Some drivers (e.g. cmipci) have their own proc entries for the +register dump, etc (e.g. /proc/asound/card*/cmipci shows the register +dump). These files would be really helpful for debugging. + +When PCM devices are available on this card, you can see directories +like pcm0p or pcm1c. They hold the PCM information for each PCM +stream. The number after 'pcm' is the PCM device number from 0, and +the last 'p' or 'c' means playback or capture direction. The files in +this subtree is described later. + +The status of MIDI I/O is found in midi* files. It shows the device +name and the received/transmitted bytes through the MIDI device. + +When the card is equipped with AC97 codecs, there are codec97#* +subdirectories (described later). + +When the OSS mixer emulation is enabled (and the module is loaded), +oss_mixer file appears here, too. This shows the current mapping of +OSS mixer elements to the ALSA control elements. You can change the +mapping by writing to this device. Read OSS-Emulation.txt for +details. + + +PCM Proc Files +-------------- + +card*/pcm*/info + The general information of this PCM device: card #, device #, + substreams, etc. + +card*/pcm*/xrun_debug + This file appears when CONFIG_SND_DEBUG=y and + CONFIG_PCM_XRUN_DEBUG=y. + This shows the status of xrun (= buffer overrun/xrun) and + invalid PCM position debug/check of ALSA PCM middle layer. + It takes an integer value, can be changed by writing to this + file, such as + + # echo 5 > /proc/asound/card0/pcm0p/xrun_debug + + The value consists of the following bit flags: + bit 0 = Enable XRUN/jiffies debug messages + bit 1 = Show stack trace at XRUN / jiffies check + bit 2 = Enable additional jiffies check + + When the bit 0 is set, 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 the bit 1 is set, the driver will show the stack trace + additionally. This may help the debugging. + + Since 2.6.30, this option can enable the hwptr check using + jiffies. This detects spontaneous invalid pointer callback + values, but can be lead to too much corrections for a (mostly + buggy) hardware that doesn't give smooth pointer updates. + This feature is enabled via the bit 2. + +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. + +card*/pcm*/sub*/xrun_injection + Triggers an XRUN to the running stream when any value is + written to this proc file. Used for fault injection. + This entry is write-only. + +AC97 Codec Information +---------------------- + +card*/codec97#*/ac97#?-? + Shows the general information of this AC97 codec chip, such as + name, capabilities, set up. + +card*/codec97#0/ac97#?-?+regs + Shows the AC97 register dump. Useful for debugging. + + When CONFIG_SND_DEBUG is enabled, you can write to this file for + changing an AC97 register directly. Pass two hex numbers. + For example, + + # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs + + +USB Audio Streams +----------------- + +card*/stream* + Shows the assignment and the current status of each audio stream + of the given card. This information is very useful for debugging. + + +HD-Audio Codecs +--------------- + +card*/codec#* + Shows the general codec information and the attribute of each + widget node. + +card*/eld#* + Available for HDMI or DisplayPort interfaces. + Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, + and describes its audio capabilities and configurations. + + Some ELD fields may be modified by doing `echo name hex_value > eld#*`. + Only do this if you are sure the HDMI sink provided value is wrong. + And if that makes your HDMI audio work, please report to us so that we + can fix it in future kernel releases. + + +Sequencer Information +--------------------- + +seq/drivers + Lists the currently available ALSA sequencer drivers. + +seq/clients + Shows the list of currently available sequencer clients and + ports. The connection status and the running status are shown + in this file, too. + +seq/queues + Lists the currently allocated/running sequencer queues. + +seq/timer + Lists the currently allocated/running sequencer timers. + +seq/oss + Lists the OSS-compatible sequencer stuffs. + + +Help For Debugging? +------------------- + +When the problem is related with PCM, first try to turn on xrun_debug +mode. This will give you the kernel messages when and where xrun +happened. + +If it's really a bug, report it with the following information: + + - the name of the driver/card, show in /proc/asound/cards + - the register dump, if available (e.g. card*/cmipci) + +when it's a PCM problem, + + - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM + sub-stream directory + +when it's a mixer problem, + + - AC97 proc files, codec97#*/* files + +for USB audio/midi, + + - output of lsusb -v + - stream* files in card directory + + +The ALSA bug-tracking system is found at: + + https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44 new file mode 100644 index 000000000..67b2ea1cc --- /dev/null +++ b/Documentation/sound/alsa/README.maya44 @@ -0,0 +1,163 @@ +NOTE: The following is the original document of Rainer's patch that the +current maya44 code based on. Some contents might be obsoleted, but I +keep here as reference -- tiwai + +---------------------------------------------------------------- + +STATE OF DEVELOPMENT: + +This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. +Development is carried out by Rainer Zimmermann (mail@lightshed.de). + +ESI provided a sample Maya44 card for the development work. + +However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing. + +This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008). + + +The following functions work, as tested by Rainer Zimmermann and Piotr Makowski: + +- playback and capture at all sampling rates +- input/output level +- crossmixing +- line/mic switch +- phantom power switch +- analogue monitor a.k.a bypass + + +The following functions *should* work, but are not fully tested: + +- Channel 3+4 analogue - S/PDIF input switching +- S/PDIF output +- all inputs/outputs on the M/IO/DIO extension card +- internal/external clock selection + + +*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.* + + +Things that do not seem to work: + +- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code). + +- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down. + + +DRIVER DETAILS: + +the following files were added: + +pci/ice1724/maya44.c - Maya44 specific code +pci/ice1724/maya44.h +pci/ice1724/ice1724.patch +pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) +i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs +include/wm8776.h + + +Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. +This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately. + + +the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree: + +wtm.h +vt1720_mobo.h +revo.h +prodigy192.h +pontis.h +phase.h +maya44.h +juli.h +aureon.h +amp.h +envy24ht.h +se.h +prodigy_hifi.h + + +*I hope this is the correct way to do things.* + + +SAMPLING RATES: + +The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture. + +As the ICE1724 chip only allows one global sampling rate, this is handled as follows: + +* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels. + +* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices. + +*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality. + + +I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic. + +The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). + + +SOUND DEVICES: + +PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0): + +hw:0,0 input - stereo, analog input 1+2 +hw:0,0 output - stereo, analog output 1+2 +hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input +hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) + + +NAMING OF MIXER CONTROLS: + +(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software). + + +PCM: (digital) output level for channel 1+2 +PCM 1: same for channel 3+4 + +Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. + Make sure this is not turned on while any other source is connected to input 1/2. + It might damage the source and/or the maya44 card. + +Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). + +Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. +Bypass 1: same for channel 3+4. + +Crossmix: cross-mixer from channels 1+2 to channels 3+4 +Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 + +IEC958 Output: switch for S/PDIF output. + This is not supported by the ESI windows driver. + S/PDIF should output the same signal as channel 3+4. [untested!] + + +Digitial output selectors: + + These switches allow a direct digital routing from the ADCs to the DACs. + Each switch determines where the digital input data to one of the DACs comes from. + They are not supported by the ESI windows driver. + For normal operation, they should all be set to "PCM out". + +H/W: Output source channel 1 +H/W 1: Output source channel 2 +H/W 2: Output source channel 3 +H/W 3: Output source channel 4 + +H/W 4 ... H/W 9: unknown function, left in to enable testing. + Possibly some of these control S/PDIF output(s). + If these turn out to be unused, they will go away in later driver versions. + +Selectable values for each of the digital output selectors are: + "PCM out" -> DAC output of the corresponding channel (default setting) + "Input 1"... + "Input 4" -> direct routing from ADC output of the selected input channel + + +-------- + +Feb 14, 2008 +Rainer Zimmermann +mail@lightshed.de + diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt new file mode 100644 index 000000000..f4b5988f4 --- /dev/null +++ b/Documentation/sound/alsa/SB-Live-mixer.txt @@ -0,0 +1,356 @@ + + Sound Blaster Live mixer / default DSP code + =========================================== + + +The EMU10K1 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K1 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) IEC958 (S/PDIF) raw PCM +-------------------------- + +This PCM device (it's the 4th PCM device (index 3!) and first subdevice +(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit +little endian streams without any modifications to the digital output +(coaxial or optical). The universal interface allows the creation of up +to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would +be easy to add support for multichannel devices to the current code, +but the conversion routines exist only for stereo (2-channel streams) +at the time. + +Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. + + +2) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + + +name='Wave Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Wave Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operates +separately (they are not inside the AC97 codec). + +name='Wave Center Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? right DAC PCM slot of the AC97 codec. + +name='Wave LFE Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? left DAC PCM slot of the AC97 codec. + +name='Wave Capture Volume',index=0 +name='Wave Capture Switch',index=0 + +These controls are used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Synth 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='Synth Capture Volume',index=0 +name='Synth 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 000000000..1b0ac06ba --- /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/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt new file mode 100644 index 000000000..0fa40679b --- /dev/null +++ b/Documentation/sound/alsa/alsa-parameters.txt @@ -0,0 +1,135 @@ + ALSA Kernel Parameters + ~~~~~~~~~~~~~~~~~~~~~~ + +See Documentation/kernel-parameters.txt for general information on +specifying module parameters. + +This document may not be entirely up to date and comprehensive. The command +"modinfo -p ${modulename}" shows a current list of all parameters of a loadable +module. Loadable modules, after being loaded into the running kernel, also +reveal their parameters in /sys/module/${modulename}/parameters/. Some of these +parameters may be changed at runtime by the command +"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". + + + snd-ad1816a= [HW,ALSA] + + snd-ad1848= [HW,ALSA] + + snd-ali5451= [HW,ALSA] + + snd-als100= [HW,ALSA] + + snd-als4000= [HW,ALSA] + + snd-azt2320= [HW,ALSA] + + snd-cmi8330= [HW,ALSA] + + snd-cmipci= [HW,ALSA] + + snd-cs4231= [HW,ALSA] + + snd-cs4232= [HW,ALSA] + + snd-cs4236= [HW,ALSA] + + snd-cs4281= [HW,ALSA] + + snd-cs46xx= [HW,ALSA] + + snd-dt019x= [HW,ALSA] + + snd-dummy= [HW,ALSA] + + snd-emu10k1= [HW,ALSA] + + snd-ens1370= [HW,ALSA] + + snd-ens1371= [HW,ALSA] + + snd-es968= [HW,ALSA] + + snd-es1688= [HW,ALSA] + + snd-es18xx= [HW,ALSA] + + snd-es1938= [HW,ALSA] + + snd-es1968= [HW,ALSA] + + snd-fm801= [HW,ALSA] + + snd-gusclassic= [HW,ALSA] + + snd-gusextreme= [HW,ALSA] + + snd-gusmax= [HW,ALSA] + + snd-hdsp= [HW,ALSA] + + snd-ice1712= [HW,ALSA] + + snd-intel8x0= [HW,ALSA] + + snd-interwave= [HW,ALSA] + + snd-interwave-stb= + [HW,ALSA] + + snd-korg1212= [HW,ALSA] + + snd-maestro3= [HW,ALSA] + + snd-mpu401= [HW,ALSA] + + snd-mtpav= [HW,ALSA] + + snd-nm256= [HW,ALSA] + + snd-opl3sa2= [HW,ALSA] + + snd-opti92x-ad1848= + [HW,ALSA] + + snd-opti92x-cs4231= + [HW,ALSA] + + snd-opti93x= [HW,ALSA] + + snd-pmac= [HW,ALSA] + + snd-rme32= [HW,ALSA] + + snd-rme96= [HW,ALSA] + + snd-rme9652= [HW,ALSA] + + snd-sb8= [HW,ALSA] + + snd-sb16= [HW,ALSA] + + snd-sbawe= [HW,ALSA] + + snd-serial= [HW,ALSA] + + snd-sgalaxy= [HW,ALSA] + + snd-sonicvibes= [HW,ALSA] + + snd-sun-amd7930= + [HW,ALSA] + + snd-sun-cs4231= [HW,ALSA] + + snd-trident= [HW,ALSA] + + snd-usb-audio= [HW,ALSA,USB] + + snd-via82xx= [HW,ALSA] + + snd-virmidi= [HW,ALSA] + + snd-wavefront= [HW,ALSA] + + snd-ymfpci= [HW,ALSA] diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt new file mode 100644 index 000000000..630c492c3 --- /dev/null +++ b/Documentation/sound/alsa/compress_offload.txt @@ -0,0 +1,234 @@ + compress_offload.txt + ===================== + Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com> + Vinod Koul <vinod.koul@linux.intel.com> + +Overview + +Since its early days, the ALSA API was defined with PCM support or +constant bitrates payloads such as IEC61937 in mind. Arguments and +returned values in frames are the norm, making it a challenge to +extend the existing API to compressed data streams. + +In recent years, audio digital signal processors (DSP) were integrated +in system-on-chip designs, and DSPs are also integrated in audio +codecs. Processing compressed data on such DSPs results in a dramatic +reduction of power consumption compared to host-based +processing. Support for such hardware has not been very good in Linux, +mostly because of a lack of a generic API available in the mainline +kernel. + +Rather than requiring a compatibility break with an API change of the +ALSA PCM interface, a new 'Compressed Data' API is introduced to +provide a control and data-streaming interface for audio DSPs. + +The design of this API was inspired by the 2-year experience with the +Intel Moorestown SOC, with many corrections required to upstream the +API in the mainline kernel instead of the staging tree and make it +usable by others. + +Requirements + +The main requirements are: + +- separation between byte counts and time. Compressed formats may have + a header per file, per frame, or no header at all. The payload size + may vary from frame-to-frame. As a result, it is not possible to + estimate reliably the duration of audio buffers when handling + compressed data. Dedicated mechanisms are required to allow for + reliable audio-video synchronization, which requires precise + reporting of the number of samples rendered at any given time. + +- Handling of multiple formats. PCM data only requires a specification + of the sampling rate, number of channels and bits per sample. In + contrast, compressed data comes in a variety of formats. Audio DSPs + may also provide support for a limited number of audio encoders and + decoders embedded in firmware, or may support more choices through + dynamic download of libraries. + +- Focus on main formats. This API provides support for the most + popular formats used for audio and video capture and playback. It is + likely that as audio compression technology advances, new formats + will be added. + +- Handling of multiple configurations. Even for a given format like + AAC, some implementations may support AAC multichannel but HE-AAC + stereo. Likewise WMA10 level M3 may require too much memory and cpu + cycles. The new API needs to provide a generic way of listing these + formats. + +- Rendering/Grabbing only. This API does not provide any means of + hardware acceleration, where PCM samples are provided back to + user-space for additional processing. This API focuses instead on + streaming compressed data to a DSP, with the assumption that the + decoded samples are routed to a physical output or logical back-end. + + - Complexity hiding. Existing user-space multimedia frameworks all + have existing enums/structures for each compressed format. This new + API assumes the existence of a platform-specific compatibility layer + to expose, translate and make use of the capabilities of the audio + DSP, eg. Android HAL or PulseAudio sinks. By construction, regular + applications are not supposed to make use of this API. + + +Design + +The new API shares a number of concepts with the PCM API for flow +control. Start, pause, resume, drain and stop commands have the same +semantics no matter what the content is. + +The concept of memory ring buffer divided in a set of fragments is +borrowed from the ALSA PCM API. However, only sizes in bytes can be +specified. + +Seeks/trick modes are assumed to be handled by the host. + +The notion of rewinds/forwards is not supported. Data committed to the +ring buffer cannot be invalidated, except when dropping all buffers. + +The Compressed Data API does not make any assumptions on how the data +is transmitted to the audio DSP. DMA transfers from main memory to an +embedded audio cluster or to a SPI interface for external DSPs are +possible. As in the ALSA PCM case, a core set of routines is exposed; +each driver implementer will have to write support for a set of +mandatory routines and possibly make use of optional ones. + +The main additions are + +- get_caps +This routine returns the list of audio formats supported. Querying the +codecs on a capture stream will return encoders, decoders will be +listed for playback streams. + +- get_codec_caps For each codec, this routine returns a list of +capabilities. The intent is to make sure all the capabilities +correspond to valid settings, and to minimize the risks of +configuration failures. For example, for a complex codec such as AAC, +the number of channels supported may depend on a specific profile. If +the capabilities were exposed with a single descriptor, it may happen +that a specific combination of profiles/channels/formats may not be +supported. Likewise, embedded DSPs have limited memory and cpu cycles, +it is likely that some implementations make the list of capabilities +dynamic and dependent on existing workloads. In addition to codec +settings, this routine returns the minimum buffer size handled by the +implementation. This information can be a function of the DMA buffer +sizes, the number of bytes required to synchronize, etc, and can be +used by userspace to define how much needs to be written in the ring +buffer before playback can start. + +- set_params +This routine sets the configuration chosen for a specific codec. The +most important field in the parameters is the codec type; in most +cases decoders will ignore other fields, while encoders will strictly +comply to the settings + +- get_params +This routines returns the actual settings used by the DSP. Changes to +the settings should remain the exception. + +- get_timestamp +The timestamp becomes a multiple field structure. It lists the number +of bytes transferred, the number of samples processed and the number +of samples rendered/grabbed. All these values can be used to determine +the average bitrate, figure out if the ring buffer needs to be +refilled or the delay due to decoding/encoding/io on the DSP. + +Note that the list of codecs/profiles/modes was derived from the +OpenMAX AL specification instead of reinventing the wheel. +Modifications include: +- Addition of FLAC and IEC formats +- Merge of encoder/decoder capabilities +- Profiles/modes listed as bitmasks to make descriptors more compact +- Addition of set_params for decoders (missing in OpenMAX AL) +- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL) +- Addition of format information for WMA +- Addition of encoding options when required (derived from OpenMAX IL) +- Addition of rateControlSupported (missing in OpenMAX AL) + +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + +Not supported: + +- Support for VoIP/circuit-switched calls is not the target of this + API. Support for dynamic bit-rate changes would require a tight + coupling between the DSP and the host stack, limiting power savings. + +- Packet-loss concealment is not supported. This would require an + additional interface to let the decoder synthesize data when frames + are lost during transmission. This may be added in the future. + +- Volume control/routing is not handled by this API. Devices exposing a + compressed data interface will be considered as regular ALSA devices; + volume changes and routing information will be provided with regular + ALSA kcontrols. + +- Embedded audio effects. Such effects should be enabled in the same + manner, no matter if the input was PCM or compressed. + +- multichannel IEC encoding. Unclear if this is required. + +- Encoding/decoding acceleration is not supported as mentioned + above. It is possible to route the output of a decoder to a capture + stream, or even implement transcoding capabilities. This routing + would be enabled with ALSA kcontrols. + +- Audio policy/resource management. This API does not provide any + hooks to query the utilization of the audio DSP, nor any preemption + mechanisms. + +- No notion of underrun/overrun. Since the bytes written are compressed + in nature and data written/read doesn't translate directly to + rendered output in time, this does not deal with underrun/overrun and + maybe dealt in user-library + +Credits: +- Mark Brown and Liam Girdwood for discussions on the need for this API +- Harsha Priya for her work on intel_sst compressed API +- Rakesh Ughreja for valuable feedback +- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for + demonstrating and quantifying the benefits of audio offload on a + real platform. diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt new file mode 100644 index 000000000..751d45036 --- /dev/null +++ b/Documentation/sound/alsa/emu10k1-jack.txt @@ -0,0 +1,74 @@ +This document is a guide to using the emu10k1 based devices with JACK for low +latency, multichannel recording functionality. All of my recent work to allow +Linux users to use the full capabilities of their hardware has been inspired +by the kX Project. Without their work I never would have discovered the true +power of this hardware. + + http://www.kxproject.com + - Lee Revell, 2005.03.30 + +Low latency, multichannel audio with JACK and the emu10k1/emu10k2 +----------------------------------------------------------------- + +Until recently, emu10k1 users on Linux did not have access to the same low +latency, multichannel features offered by the "kX ASIO" feature of their +Windows driver. As of ALSA 1.0.9 this is no more! + +For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback +channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or +even 32 (0.66ms) frames should work well. + +The configuration is slightly more involved than on Windows, as you have to +select the correct device for JACK to use. Actually, for qjackctl users it's +fairly self explanatory - select Duplex, then for capture and playback select +the multichannel devices, set the in and out channels to 16, and the sample +rate to 48000Hz. The command line looks like this: + +/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S + +This will give you 16 input ports and 16 output ports. + +The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the +Audigy). The mapping from FX bus to physical output is described in +SB-Live-mixer.txt (or Audigy-mixer.txt). + +The 16 input ports are connected to the 16 physical inputs. Contrary to +popular belief, all emu10k1 cards are multichannel cards. Which of these +input channels have physical inputs connected to them depends on the card +model. Trial and error is highly recommended; the pinout diagrams +for the card have been reverse engineered by some enterprising kX users and are +available on the internet. Meterbridge is helpful here, and the kX forums are +packed with useful information. + +Each input port will either correspond to a digital (SPDIF) input, an analog +input, or nothing. The one exception is the SBLive! 5.1. On these devices, +the second and third input ports are wired to the center/LFE output. You will +still see 16 capture channels, but only 14 are available for recording inputs. + +This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK +ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output) +channels. + +/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards: +-------------------------------------------- +JACK Epilog FXBUS2(nr) +-------------------------------------------- +capture_1 asio14 FXBUS2(0xe) +capture_2 asio15 FXBUS2(0xf) +capture_3 asio0 FXBUS2(0x0) +~capture_4 Center EXTOUT(0x11) // mapped to by Center +~capture_5 LFE EXTOUT(0x12) // mapped to by LFE +capture_6 asio3 FXBUS2(0x3) +capture_7 asio4 FXBUS2(0x4) +capture_8 asio5 FXBUS2(0x5) +capture_9 asio6 FXBUS2(0x6) +capture_10 asio7 FXBUS2(0x7) +capture_11 asio8 FXBUS2(0x8) +capture_12 asio9 FXBUS2(0x9) +capture_13 asio10 FXBUS2(0xa) +capture_14 asio11 FXBUS2(0xb) +capture_15 asio12 FXBUS2(0xc) +capture_16 asio13 FXBUS2(0xd) +*/ + +TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt new file mode 100644 index 000000000..de8efbc7e --- /dev/null +++ b/Documentation/sound/alsa/hda_codec.txt @@ -0,0 +1,322 @@ +Notes on Universal Interface for Intel High Definition Audio Codec +------------------------------------------------------------------ + +Takashi Iwai <tiwai@suse.de> + + +[Still a draft version] + + +General +======= + +The snd-hda-codec module supports the generic access function for the +High Definition (HD) audio codecs. It's designed to be independent +from the controller code like ac97 codec module. The real accessors +from/to the controller must be implemented in the lowlevel driver. + +The structure of this module is similar with ac97_codec module. +Each codec chip belongs to a bus class which communicates with the +controller. + + +Initialization of Bus Instance +============================== + +The card driver has to create struct hda_bus at first. The template +struct should be filled and passed to the constructor: + +struct hda_bus_template { + void *private_data; + struct pci_dev *pci; + const char *modelname; + struct hda_bus_ops ops; +}; + +The card driver can set and use the private_data field to retrieve its +own data in callback functions. The pci field is used when the patch +needs to check the PCI subsystem IDs, so on. For non-PCI system, it +doesn't have to be set, of course. +The modelname field specifies the board's specific configuration. The +string is passed to the codec parser, and it depends on the parser how +the string is used. +These fields, private_data, pci and modelname are all optional. + +The ops field contains the callback functions as the following: + +struct hda_bus_ops { + int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); + unsigned int (*get_response)(struct hda_codec *codec); + void (*private_free)(struct hda_bus *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*pm_notify)(struct hda_codec *codec); +#endif +}; + +The command callback is called when the codec module needs to send a +VERB to the controller. It's always a single command. +The get_response callback is called when the codec requires the answer +for the last command. These two callbacks are mandatory and have to +be given. +The third, private_free callback, is optional. It's called in the +destructor to release any necessary data in the lowlevel driver. + +The pm_notify callback is available only with +CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs +to power up or may power down. The controller should check the all +belonging codecs on the bus whether they are actually powered off +(check codec->power_on), and optionally the driver may power down the +controller side, too. + +The bus instance is created via snd_hda_bus_new(). You need to pass +the card instance, the template, and the pointer to store the +resultant bus instance. + +int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, + struct hda_bus **busp); + +It returns zero if successful. A negative return value means any +error during creation. + + +Creation of Codec Instance +========================== + +Each codec chip on the board is then created on the BUS instance. +To create a codec instance, call snd_hda_codec_new(). + +int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + struct hda_codec **codecp); + +The first argument is the BUS instance, the second argument is the +address of the codec, and the last one is the pointer to store the +resultant codec instance (can be NULL if not needed). + +The codec is stored in a linked list of bus instance. You can follow +the codec list like: + + struct hda_codec *codec; + list_for_each_entry(codec, &bus->codec_list, list) { + ... + } + +The codec isn't initialized at this stage properly. The +initialization sequence is called when the controls are built later. + + +Codec Access +============ + +To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). +snd_hda_param_read() is for reading parameters. +For writing a sequence of verbs, use snd_hda_sequence_write(). + +There are variants of cached read/write, snd_hda_codec_write_cache(), +snd_hda_sequence_write_cache(). These are used for recording the +register states for the power-management resume. When no PM is needed, +these are equivalent with non-cached version. + +To retrieve the number of sub nodes connected to the given node, use +snd_hda_get_sub_nodes(). The connection list can be obtained via +snd_hda_get_connections() call. + +When an unsolicited event happens, pass the event via +snd_hda_queue_unsol_event() so that the codec routines will process it +later. + + +(Mixer) Controls +================ + +To create mixer controls of all codecs, call +snd_hda_build_controls(). It then builds the mixers and does +initialization stuff on each codec. + + +PCM Stuff +========= + +snd_hda_build_pcms() gives the necessary information to create PCM +streams. When it's called, each codec belonging to the bus stores +codec->num_pcms and codec->pcm_info fields. The num_pcms indicates +the number of elements in pcm_info array. The card driver is supposed +to traverse the codec linked list, read the pcm information in +pcm_info array, and build pcm instances according to them. + +The pcm_info array contains the following record: + +/* PCM information for each substream */ +struct hda_pcm_stream { + unsigned int substreams; /* number of substreams, 0 = not exist */ + unsigned int channels_min; /* min. number of channels */ + unsigned int channels_max; /* max. number of channels */ + hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ + u32 rates; /* supported rates */ + u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ + unsigned int maxbps; /* supported max. bit per sample */ + struct hda_pcm_ops ops; +}; + +/* for PCM creation */ +struct hda_pcm { + char *name; + struct hda_pcm_stream stream[2]; +}; + +The name can be passed to snd_pcm_new(). The stream field contains +the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and +capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver +should pass substreams to snd_pcm_new() for the number of substreams +to create. + +The channels_min, channels_max, rates and formats should be copied to +runtime->hw record. They and maxbps fields are used also to compute +the format value for the HDA codec and controller. Call +snd_hda_calc_stream_format() to get the format value. + +The ops field contains the following callback functions: + +struct hda_pcm_ops { + int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, + unsigned int stream_tag, unsigned int format, + struct snd_pcm_substream *substream); + int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); +}; + +All are non-NULL, so you can call them safely without NULL check. + +The open callback should be called in PCM open after runtime->hw is +set up. It may override some setting and constraints additionally. +Similarly, the close callback should be called in the PCM close. + +The prepare callback should be called in PCM prepare. This will set +up the codec chip properly for the operation. The cleanup should be +called in hw_free to clean up the configuration. + +The caller should check the return value, at least for open and +prepare callbacks. When a negative value is returned, some error +occurred. + + +Proc Files +========== + +Each codec dumps the widget node information in +/proc/asound/card*/codec#* file. This information would be really +helpful for debugging. Please provide its contents together with the +bug report. + + +Power Management +================ + +It's simple: +Call snd_hda_suspend() in the PM suspend callback. +Call snd_hda_resume() in the PM resume callback. + + +Codec Preset (Patch) +==================== + +To set up and handle the codec functionality fully, each codec may +have a codec preset (patch). It's defined in struct hda_codec_preset: + + struct hda_codec_preset { + unsigned int id; + unsigned int mask; + unsigned int subs; + unsigned int subs_mask; + unsigned int rev; + const char *name; + int (*patch)(struct hda_codec *codec); + }; + +When the codec id and codec subsystem id match with the given id and +subs fields bitwise (with bitmask mask and subs_mask), the callback +patch is called. The patch callback should initialize the codec and +set the codec->patch_ops field. This is defined as below: + + struct hda_codec_ops { + int (*build_controls)(struct hda_codec *codec); + int (*build_pcms)(struct hda_codec *codec); + int (*init)(struct hda_codec *codec); + void (*free)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + #ifdef CONFIG_PM + int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*resume)(struct hda_codec *codec); + #endif + #ifdef CONFIG_SND_HDA_POWER_SAVE + int (*check_power_status)(struct hda_codec *codec, + hda_nid_t nid); + #endif + }; + +The build_controls callback is called from snd_hda_build_controls(). +Similarly, the build_pcms callback is called from +snd_hda_build_pcms(). The init callback is called after +build_controls to initialize the hardware. +The free callback is called as a destructor. + +The unsol_event callback is called when an unsolicited event is +received. + +The suspend and resume callbacks are for power management. +They can be NULL if no special sequence is required. When the resume +callback is NULL, the driver calls the init callback and resumes the +registers from the cache. If other handling is needed, you'd need to +write your own resume callback. There, the amp values can be resumed +via + void snd_hda_codec_resume_amp(struct hda_codec *codec); +and the other codec registers via + void snd_hda_codec_resume_cache(struct hda_codec *codec); + +The check_power_status callback is called when the amp value of the +given widget NID is changed. The codec code can turn on/off the power +appropriately from this information. + +Each entry can be NULL if not necessary to be called. + + +Generic Parser +============== + +When the device doesn't match with any given presets, the widgets are +parsed via th generic parser (hda_generic.c). Its support is +limited: no multi-channel support, for example. + + +Digital I/O +=========== + +Call snd_hda_create_spdif_out_ctls() from the patch to create controls +related with SPDIF out. + + +Helper Functions +================ + +snd_hda_get_codec_name() stores the codec name on the given string. + +snd_hda_check_board_config() can be used to obtain the configuration +information matching with the device. Define the model string table +and the table with struct snd_pci_quirk entries (zero-terminated), +and pass it to the function. The function checks the modelname given +as a module parameter, and PCI subsystem IDs. If the matching entry +is found, it returns the config field value. + +snd_hda_add_new_ctls() can be used to create and add control entries. +Pass the zero-terminated array of struct snd_kcontrol_new + +Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be +used for the entry of struct snd_kcontrol_new. + +The input MUX helper callbacks for such a control are provided, too: +snd_hda_input_mux_info() and snd_hda_input_mux_put(). See +patch_realtek.c for example. diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt new file mode 100644 index 000000000..7ba31948d --- /dev/null +++ b/Documentation/sound/alsa/hdspm.txt @@ -0,0 +1,362 @@ +Software Interface ALSA-DSP MADI Driver + +(translated from German, so no good English ;-), +2004 - winfried ritsch + + + + Full functionality has been added to the driver. Since some of + the Controls and startup-options are ALSA-Standard and only the + special Controls are described and discussed below. + + + hardware functionality: + + + Audio transmission: + + number of channels -- depends on transmission mode + + The number of channels chosen is from 1..Nmax. The reason to + use for a lower number of channels is only resource allocation, + since unused DMA channels are disabled and less memory is + allocated. So also the throughput of the PCI system can be + scaled. (Only important for low performance boards). + + Single Speed -- 1..64 channels + + (Note: Choosing the 56channel mode for transmission or as + receiver, only 56 are transmitted/received over the MADI, but + all 64 channels are available for the mixer, so channel count + for the driver) + + Double Speed -- 1..32 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 28 are transmitted/received + over the MADI, but all 32 channels are available for the mixer, + so channel count for the driver + + + Quad Speed -- 1..16 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 14 are transmitted/received + over the MADI, but all 16 channels are available for the mixer, + so channel count for the driver + + Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) + + Sample Rates -- + + Single Speed -- 32000, 44100, 48000 + + Double Speed -- 64000, 88200, 96000 (untested) + + Quad Speed -- 128000, 176400, 192000 (untested) + + access-mode -- MMAP (memory mapped), Not interleaved + (PCM_NON-INTERLEAVED) + + buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples + + fragments -- 2 + + Hardware-pointer -- 2 Modi + + + The Card supports the readout of the actual Buffer-pointer, + where DMA reads/writes. Since of the bulk mode of PCI it is only + 64 Byte accurate. SO it is not really usable for the + ALSA-mid-level functions (here the buffer-ID gives a better + result), but if MMAP is used by the application. Therefore it + can be configured at load-time with the parameter + precise-pointer. + + + (Hint: Experimenting I found that the pointer is maximum 64 to + large never to small. So if you subtract 64 you always have a + safe pointer for writing, which is used on this mode inside + ALSA. In theory now you can get now a latency as low as 16 + Samples, which is a quarter of the interrupt possibilities.) + + Precise Pointer -- off + interrupt used for pointer-calculation + + Precise Pointer -- on + hardware pointer used. + + Controller: + + + Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to + use the standard mixer-controls, since this would break most of + (especially graphic) ALSA-Mixer GUIs. So Mixer control has be + provided by a 2-dimensional controller using the + hwdep-interface. + + Also all 128+256 Peak and RMS-Meter can be accessed via the + hwdep-interface. Since it could be a performance problem always + copying and converting Peak and RMS-Levels even if you just need + one, I decided to export the hardware structure, so that of + needed some driver-guru can implement a memory-mapping of mixer + or peak-meters over ioctl, or also to do only copying and no + conversion. A test-application shows the usage of the controller. + + Latency Controls --- not implemented !!! + + + Note: Within the windows-driver the latency is accessible of a + control-panel, but buffer-sizes are controlled with ALSA from + hwparams-calls and should not be changed in run-state, I did not + implement it here. + + + System Clock -- suspended !!!! + + Name -- "System Clock Mode" + + Access -- Read Write + + Values -- "Master" "Slave" + + + !!!! This is a hardware-function but is in conflict with the + Clock-source controller, which is a kind of ALSA-standard. I + makes sense to set the card to a special mode (master at some + frequency or slave), since even not using an Audio-application + a studio should have working synchronisations setup. So use + Clock-source-controller instead !!!! + + Clock Source + + Name -- "Sample Clock Source" + + Access -- Read Write + + Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", + "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", + "Internal 96.0 kHz" + + Choose between Master at a specific Frequency and so also the + Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref" + + + !!!! This is no pure hardware function but was implemented by + ALSA by some ALSA-drivers before, so I use it also. !!! + + + Preferred Sync Ref + + Name -- "Preferred Sync Reference" + + Access -- Read Write + + Values -- "Word" "MADI" + + + Within the Auto-sync-Mode the preferred Sync Source can be + chosen. If it is not available another is used if possible. + + Note: Since MADI has a much higher bit-rate than word-clock, the + card should synchronise better in MADI Mode. But since the + RME-PLL is very good, there are almost no problems with + word-clock too. I never found a difference. + + + TX 64 channel --- + + Name -- "TX 64 channels mode" + + Access -- Read Write + + Values -- 0 1 + + Using 64-channel-modus (1) or 56-channel-modus for + MADI-transmission (0). + + + Note: This control is for output only. Input-mode is detected + automatically from hardware sending MADI. + + + Clear TMS --- + + Name -- "Clear Track Marker" + + Access -- Read Write + + Values -- 0 1 + + + Don't use to lower 5 Audio-bits on AES as additional Bits. + + + Safe Mode oder Auto Input --- + + Name -- "Safe Mode" + + Access -- Read Write + + Values -- 0 1 + + (default on) + + If on (1), then if either the optical or coaxial connection + has a failure, there is a takeover to the working one, with no + sample failure. Its only useful if you use the second as a + backup connection. + + Input --- + + Name -- "Input Select" + + Access -- Read Write + + Values -- optical coaxial + + + Choosing the Input, optical or coaxial. If Safe-mode is active, + this is the preferred Input. + +-------------- Mixer ---------------------- + + Mixer + + Name -- "Mixer" + + Access -- Read Write + + Values - <channel-number 0-127> <Value 0-65535> + + + Here as a first value the channel-index is taken to get/set the + corresponding mixer channel, where 0-63 are the input to output + fader and 64-127 the playback to outputs fader. Value 0 + is channel muted 0 and 32768 an amplification of 1. + + Chn 1-64 + + fast mixer for the ALSA-mixer utils. The diagonal of the + mixer-matrix is implemented from playback to output. + + + Line Out + + Name -- "Line Out" + + Access -- Read Write + + Values -- 0 1 + + Switching on and off the analog out, which has nothing to do + with mixing or routing. the analog outs reflects channel 63,64. + + +--- information (only read access): + + Sample Rate + + Name -- "System Sample Rate" + + Access -- Read-only + + getting the sample rate. + + + External Rate measured + + Name -- "External Rate" + + Access -- Read only + + + Should be "Autosync Rate", but Name used is + ALSA-Scheme. External Sample frequency liked used on Autosync is + reported. + + + MADI Sync Status + + Name -- "MADI Sync Lock Status" + + Access -- Read + + Values -- 0,1,2 + + MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. + + + Word Clock Sync Status + + Name -- "Word Clock Lock Status" + + Access -- Read + + Values -- 0,1,2 + + Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. + + AutoSync + + Name -- "AutoSync Reference" + + Access -- Read + + Values -- "WordClock", "MADI", "None" + + Sync-Reference is either "WordClock", "MADI" or none. + + RX 64ch --- noch nicht implementiert + + MADI-Receiver is in 64 channel mode oder 56 channel mode. + + + AB_inp --- not tested + + Used input for Auto-Input. + + + actual Buffer Position --- not implemented + + !!! this is a ALSA internal function, so no control is used !!! + + + +Calling Parameter: + + index int array (min = 1, max = 8), + "Index value for RME HDSPM interface." card-index within ALSA + + note: ALSA-standard + + id string array (min = 1, max = 8), + "ID string for RME HDSPM interface." + + note: ALSA-standard + + enable int array (min = 1, max = 8), + "Enable/disable specific HDSPM sound-cards." + + note: ALSA-standard + + precise_ptr int array (min = 1, max = 8), + "Enable precise pointer, or disable." + + note: Use only when the application supports this (which is a special case). + + line_outs_monitor int array (min = 1, max = 8), + "Send playback streams to analog outs by default." + + + note: each playback channel is mixed to the same numbered output + channel (routed). This is against the ALSA-convention, where all + channels have to be muted on after loading the driver, but was + used before on other cards, so i historically use it again) + + + + enable_monitor int array (min = 1, max = 8), + "Enable Analog Out on Channel 63/64 by default." + + note: here the analog output is enabled (but not routed). diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt new file mode 100644 index 000000000..9657e8099 --- /dev/null +++ b/Documentation/sound/alsa/powersave.txt @@ -0,0 +1,41 @@ +Notes on Power-Saving Mode +========================== + +AC97 and HD-audio drivers have the automatic power-saving mode. +This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE +and CONFIG_SND_HDA_POWER_SAVE options, respectively. + +With the automatic power-saving, the driver turns off the codec power +appropriately when no operation is required. When no applications use +the device and/or no analog loopback is set, the power disablement is +done fully or partially. It'll save a certain power consumption, thus +good for laptops (even for desktops). + +The time-out for automatic power-off can be specified via power_save +module option of snd-ac97-codec and snd-hda-intel modules. Specify +the time-out value in seconds. 0 means to disable the automatic +power-saving. The default value of timeout is given via +CONFIG_SND_AC97_POWER_SAVE_DEFAULT and +CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 +(the minimum value) isn't recommended because many applications try to +reopen the device frequently. 10 would be a good choice for normal +operations. + +The power_save option is exported as writable. This means you can +adjust the value via sysfs on the fly. For example, to turn on the +automatic power-save mode with 10 seconds, write to +/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): + + # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + + +Note that you might hear click noise/pop when changing the power +state. Also, it often takes certain time to wake up from the +power-down to the active state. These are often hardly to fix, so +don't report extra bug reports unless you have a fix patch ;-) + +For HD-audio interface, there is another module option, +power_save_controller. This enables/disables the power-save mode of +the controller side. Setting this on may reduce a bit more power +consumption, but might result in longer wake-up time and click noise. +Try to turn it off when you experience such a thing too often. diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html new file mode 100644 index 000000000..9663b45f6 --- /dev/null +++ b/Documentation/sound/alsa/seq_oss.html @@ -0,0 +1,409 @@ +<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> +<HTML> +<HEAD> + <TITLE>OSS Sequencer Emulation on ALSA</TITLE> +</HEAD> +<BODY> + +<CENTER> +<H1> + +<HR WIDTH="100%"></H1></CENTER> + +<CENTER> +<H1> +OSS Sequencer Emulation on ALSA</H1></CENTER> + +<HR WIDTH="100%"> +<P>Copyright (c) 1998,1999 by Takashi Iwai +<TT><A HREF="mailto:iwai@ww.uni-erlangen.de"><iwai@ww.uni-erlangen.de></A></TT> +<P>ver.0.1.8; Nov. 16, 1999 +<H2> + +<HR WIDTH="100%"></H2> + +<H2> +1. Description</H2> +This directory contains the OSS sequencer emulation driver on ALSA. Note +that this program is still in the development state. +<P>What this does - it provides the emulation of the OSS sequencer, access +via +<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices. +The most of applications using OSS can run if the appropriate ALSA +sequencer is prepared. +<P>The following features are emulated by this driver: +<UL> +<LI> +Normal sequencer and MIDI events:</LI> + +<BR>They are converted to the ALSA sequencer events, and sent to the corresponding +port. +<LI> +Timer events:</LI> + +<BR>The timer is not selectable by ioctl. The control rate is fixed to +100 regardless of HZ. That is, even on Alpha system, a tick is always +1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>. + +<LI> +Patch loading:</LI> + +<BR>It purely depends on the synth drivers whether it's supported since +the patch loading is realized by callback to the synth driver. +<LI> +I/O controls:</LI> + +<BR>Most of controls are accepted. Some controls +are dependent on the synth driver, as well as even on original OSS.</UL> +Furthermore, you can find the following advanced features: +<UL> +<LI> +Better queue mechanism:</LI> + +<BR>The events are queued before processing them. +<LI> +Multiple applications:</LI> + +<BR>You can run two or more applications simultaneously (even for OSS sequencer)! +However, each MIDI device is exclusive - that is, if a MIDI device is opened +once by some application, other applications can't use it. No such a restriction +in synth devices. +<LI> +Real-time event processing:</LI> + +<BR>The events can be processed in real time without using out of bound +ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed +events will be processed in real-time without queued. To switch off the +real-time mode, send RELTIME 0 event. +<LI> +<TT>/proc</TT> interface:</LI> + +<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT> +at any time. In the later version, configuration will be changed via <TT>/proc</TT> +interface, too.</UL> + +<H2> +2. Installation</H2> +Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>) +and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT> +will be created. If the synth module of your sound card supports for OSS +emulation (so far, only Emu8000 driver), this module will be loaded automatically. +Otherwise, you need to load this module manually. +<P>At beginning, this module probes all the MIDI ports which have been +already connected to the sequencer. Once after that, the creation and deletion +of ports are watched by announcement mechanism of ALSA sequencer. +<P>The available synth and MIDI devices can be found in proc interface. +Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example, +if you use an AWE64 card, you'll see like the following: +<PRE> OSS sequencer emulation version 0.1.8 + ALSA client number 63 + ALSA receiver port 0 + + Number of applications: 0 + + Number of synth devices: 1 + + synth 0: [EMU8000] + type 0x1 : subtype 0x20 : voices 32 + capabilties : ioctl enabled / load_patch enabled + + Number of MIDI devices: 3 + + midi 0: [Emu8000 Port-0] ALSA port 65:0 + capability write / opened none + + midi 1: [Emu8000 Port-1] ALSA port 65:1 + capability write / opened none + + midi 2: [0: MPU-401 (UART)] ALSA port 64:0 + capability read/write / opened none</PRE> +Note that the device number may be different from the information of +<TT>/proc/asound/oss-devices</TT> +or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT> +to play via OSS sequencer emulation. +<H2> +3. Using Synthesizer Devices</H2> +Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 +and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload, +too. +<P>If the lowlevel driver supports multiple access to synth devices (like +Emu8000 driver), two or more applications are allowed to run at the same +time. +<H2> +4. Using MIDI Devices</H2> +So far, only MIDI output was tested. MIDI input was not checked at all, +but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>. +Be aware that these numbers are mostly different from the list in +<TT>/proc/asound/oss-devices</TT>. +<H2> +5. Module Options</H2> +The following module options are available: +<UL> +<LI> +<TT>maxqlen</TT></LI> + +<BR>specifies the maximum read/write queue length. This queue is private +for OSS sequencer, so that it is independent from the queue length of ALSA +sequencer. Default value is 1024. +<LI> +<TT>seq_oss_debug</TT></LI> + +<BR>specifies the debug level and accepts zero (= no debug message) or +positive integer. Default value is 0.</UL> + +<H2> +6. Queue Mechanism</H2> +OSS sequencer emulation uses an ALSA priority queue. The +events from <TT>/dev/sequencer</TT> are processed and put onto the queue +specified by module option. +<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning. +The timing events are also parsed at this moment, so that the events may +be processed in real-time. Sending an event ABSTIME 0 switches the operation +mode to real-time mode, and sending an event RELTIME 0 switches it off. +In the real-time mode, all events are dispatched immediately. +<P>The queued events are dispatched to the corresponding ALSA sequencer +ports after scheduled time by ALSA sequencer dispatcher. +<P>If the write-queue is full, the application sleeps until a certain amount +(as default one half) becomes empty in blocking mode. The synchronization +to write timing was implemented, too. +<P>The input from MIDI devices or echo-back events are stored on read FIFO +queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the +process will be awaked. + +<H2> +7. Interface to Synthesizer Device</H2> + +<H3> +7.1. Registration</H3> +To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT> +function. +<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, + snd_seq_oss_callback_t *oper, void *private_data)</PRE> +The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and +<TT>nvoices</TT> +are used for making the appropriate synth_info structure for ioctl. The +return value is an index number of this device. This index must be remembered +for unregister. If registration is failed, -errno will be returned. +<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>: +<PRE>int snd_seq_oss_synth_unregister(int index),</PRE> +where the <TT>index</TT> is the index number returned by register function. +<H3> +7.2. Callbacks</H3> +OSS synthesizer devices have capability for sample downloading and ioctls +like sample reset. In OSS emulation, these special features are realized +by using callbacks. The registration argument oper is used to specify these +callbacks. The following callback functions must be defined: +<PRE>snd_seq_oss_callback_t: + int (*open)(snd_seq_oss_arg_t *p, void *closure); + int (*close)(snd_seq_oss_arg_t *p); + int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); + int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); + int (*reset)(snd_seq_oss_arg_t *p); +Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed +to be NULL. +<P>Each callback function takes the argument type snd_seq_oss_arg_t as the +first argument. +<PRE>struct snd_seq_oss_arg_t { + int app_index; + int file_mode; + int seq_mode; + snd_seq_addr_t addr; + void *private_data; + int event_passing; +};</PRE> +The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and +<TT>seq_mode</TT> +are initialized by OSS sequencer. The <TT>app_index</TT> is the application +index which is unique to each application opening OSS sequencer. The +<TT>file_mode</TT> +is bit-flags indicating the file operation mode. See +<TT>seq_oss.h</TT> +for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In +the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used. +<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be +filled by the synth driver at open callback. The <TT>addr</TT> contains +the address of ALSA sequencer port which is assigned to this device. If +the driver allocates memory for <TT>private_data</TT>, it must be released +in close callback by itself. +<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on +/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded +as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT> +mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT> +mode checks the note above 128 and regards it as key pressure event (mainly +for Emu8000 driver). +<H4> +7.2.1. Open Callback</H4> +The <TT>open</TT> is called at each time this device is opened by an application +using OSS sequencer. This must not be NULL. Typically, the open callback +does the following procedure: +<OL> +<LI> +Allocate private data record.</LI> + +<LI> +Create an ALSA sequencer port.</LI> + +<LI> +Set the new port address on arg->addr.</LI> + +<LI> +Set the private data record pointer on arg->private_data.</LI> +</OL> +Note that the type bit-flags in port_info of this synth port must NOT contain +<TT>TYPE_MIDI_GENERIC</TT> +bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT> +bit should NOT be included, too. This is necessary to tell it from other +normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, +return -errno. +<H4> +7.2.2 Ioctl Callback</H4> +The <TT>ioctl</TT> callback is called when the sequencer receives device-specific +ioctls. The following two ioctls should be processed by this callback: +<UL> +<LI> +<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI> + +<BR>reset all samples on memory -- return 0 +<LI> +<TT>IOCTL_SYNTH_MEMAVL</TT></LI> + +<BR>return the available memory size +<LI> +<TT>FM_4OP_ENABLE</TT></LI> + +<BR>can be ignored usually</UL> +The other ioctls are processed inside the sequencer without passing to +the lowlevel driver. +<H4> +7.2.3 Load_Patch Callback</H4> +The <TT>load_patch</TT> callback is used for sample-downloading. This callback +must read the data on user-space and transfer to each device. Return 0 +if succeeded, and -errno if failed. The format argument is the patch key +in patch_info record. The buf is user-space pointer where patch_info record +is stored. The offs can be ignored. The count is total data size of this +sample data. +<H4> +7.2.4 Close Callback</H4> +The <TT>close</TT> callback is called when this device is closed by the +application. If any private data was allocated in open callback, it must +be released in the close callback. The deletion of ALSA port should be +done here, too. This callback must not be NULL. +<H4> +7.2.5 Reset Callback</H4> +The <TT>reset</TT> callback is called when sequencer device is reset or +closed by applications. The callback should turn off the sounds on the +relevant port immediately, and initialize the status of the port. If this +callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the +port. +<H3> +7.3 Events</H3> +Most of the events are processed by sequencer and translated to the adequate +ALSA sequencer events, so that each synth device can receive by input_event +callback of ALSA sequencer port. The following ALSA events should be implemented +by the driver: +<BR> +<TABLE BORDER WIDTH="75%" NOSAVE > +<TR NOSAVE> +<TD NOSAVE><B>ALSA event</B></TD> + +<TD><B>Original OSS events</B></TD> +</TR> + +<TR> +<TD>NOTEON</TD> + +<TD>SEQ_NOTEON +<BR>MIDI_NOTEON</TD> +</TR> + +<TR> +<TD>NOTE</TD> + +<TD>SEQ_NOTEOFF +<BR>MIDI_NOTEOFF</TD> +</TR> + +<TR NOSAVE> +<TD NOSAVE>KEYPRESS</TD> + +<TD>MIDI_KEY_PRESSURE</TD> +</TR> + +<TR NOSAVE> +<TD>CHANPRESS</TD> + +<TD NOSAVE>SEQ_AFTERTOUCH +<BR>MIDI_CHN_PRESSURE</TD> +</TR> + +<TR NOSAVE> +<TD NOSAVE>PGMCHANGE</TD> + +<TD NOSAVE>SEQ_PGMCHANGE +<BR>MIDI_PGM_CHANGE</TD> +</TR> + +<TR> +<TD>PITCHBEND</TD> + +<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER) +<BR>MIDI_PITCH_BEND</TD> +</TR> + +<TR> +<TD>CONTROLLER</TD> + +<TD>MIDI_CTL_CHANGE +<BR>SEQ_BALANCE (with CTL_PAN)</TD> +</TR> + +<TR> +<TD>CONTROL14</TD> + +<TD>SEQ_CONTROLLER</TD> +</TR> + +<TR> +<TD>REGPARAM</TD> + +<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD> +</TR> + +<TR> +<TD>SYSEX</TD> + +<TD>SEQ_SYSEX</TD> +</TR> +</TABLE> + +<P>The most of these behavior can be realized by MIDI emulation driver +included in the Emu8000 lowlevel driver. In the future release, this module +will be independent. +<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event +type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte +packets without any modification. The lowlevel driver should process these +events appropriately. +<H2> +8. Interface to MIDI Device</H2> +Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer +ports automatically by receiving announcement from ALSA sequencer, the +MIDI devices don't need to be registered explicitly like synth devices. +However, the MIDI port_info registered to ALSA sequencer must include a group +name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or +<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>, +must be defined, too. If these conditions are not satisfied, the port is not +registered as OSS sequencer MIDI device. +<P>The events via MIDI devices are parsed in OSS sequencer and converted +to the corresponding ALSA sequencer events. The input from MIDI sequencer +is also converted to MIDI byte events by OSS sequencer. This works just +a reverse way of seq_midi module. +<H2> +9. Known Problems / TODO's</H2> + +<UL> +<LI> +Patch loading via ALSA instrument layer is not implemented yet.</LI> +</UL> + +</BODY> +</HTML> diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt new file mode 100644 index 000000000..c1919559d --- /dev/null +++ b/Documentation/sound/alsa/serial-u16550.txt @@ -0,0 +1,88 @@ + + Serial UART 16450/16550 MIDI driver + =================================== + +The adaptor module parameter allows you to select either: + + 0 - Roland Soundcanvas support (default) + 1 - Midiator MS-124T support (1) + 2 - Midiator MS-124W S/A mode (2) + 3 - MS-124W M/B mode support (3) + 4 - Generic device with multiple input support (4) + +For the Midiator MS-124W, you must set the physical M-S and A-B +switches on the Midiator to match the driver mode you select. + +In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported +(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver +sends the nonstandard MIDI command sequence F5 NN, where NN is the substream +number plus 1. Roland modules use this command to switch between different +"parts", so this feature lets you treat each part as a distinct raw MIDI +substream. The driver provides no way to send F5 00 (no selection) or to not +send the F5 NN command sequence at all; perhaps it ought to. + +Usage example for simple serial converter: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200 + +Usage example for Roland SoundCanvas with 4 MIDI ports: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4 + +In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs +module parameter is automatically set to 1. The driver sends the same data to +all four MIDI Out connectors. Set the A-B switch and the speed module +parameter to match (A=19200, B=9600). + +Usage example for MS-124T, with A-B switch in A position: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \ + speed=19200 + +In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0); +the outs module parameter is automatically set to 1. The driver sends +the same data to all four MIDI Out connectors at full MIDI speed. + +Usage example for S/A mode: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2 + +In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams; +the outs module parameter is automatically set to 16. The substream +number gives a bitmask of which MIDI Out connectors the data should be +sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to +Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports. +As a special case, midiCnD0 also sends to all ports, since it is not useful +to send the data to no ports. M/B mode has extra overhead to select the MIDI +Out for each byte, so the aggregate data rate across all four MIDI Outs is +at most one byte every 520 us, as compared with the full MIDI data rate of +one byte every 320 us per port. + +Usage example for M/B mode: + + /sbin/setserial /dev/ttyS0 uart none + /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3 + +The MS-124W hardware's M/A mode is currently not supported. This mode allows +the MIDI Outs to act independently at double the aggregate throughput of M/B, +but does not allow sending the same byte simultaneously to multiple MIDI Outs. +The M/A protocol requires the driver to twiddle the modem control lines under +timing constraints, so it would be a bit more complicated to implement than +the other modes. + +Midiator models other than MS-124W and MS-124T are currently not supported. +Note that the suffix letter is significant; the MS-124 and MS-124B are not +compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114. +I do have documentation (tim.mann@compaq.com) that partially covers these models, +but no units to experiment with. The MS-124W support is tested with a real unit. +The MS-124T support is untested, but should work. + +The Generic driver supports multiple input and output substreams over a single +serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the +appropriate input or output stream (depending on the data direction). +Additionally, the CTS signal is used to regulate the data flow. The number of +inputs is specified by the ins parameter. diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt new file mode 100644 index 000000000..c9679264c --- /dev/null +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -0,0 +1,56 @@ +ASoC currently supports the three main Digital Audio Interfaces (DAI) found on +SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. + + +AC97 +==== + + AC97 is a five wire interface commonly found on many PC sound cards. It is +now also popular in many portable devices. This DAI has a reset line and time +multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. +The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the +frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 +frame is 21uS long and is divided into 13 time slots. + +The AC97 specification can be found at :- +http://www.intel.com/p/en_US/business/design + + +I2S +=== + + I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and +Rx lines are used for audio transmission, whilst the bit clock (BCLK) and +left/right clock (LRC) synchronise the link. I2S is flexible in that either the +controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock +usually varies depending on the sample rate and the master system clock +(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate +ADC and DAC LRCLKs, this allows for simultaneous capture and playback at +different sample rates. + +I2S has several different operating modes:- + + o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC + transition. + + o Left Justified - MSB is transmitted on transition of LRC. + + o Right Justified - MSB is transmitted sample size BCLKs before LRC + transition. + +PCM +=== + +PCM is another 4 wire interface, very similar to I2S, which can support a more +flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used +to synchronise the link whilst the Tx and Rx lines are used to transmit and +receive the audio data. Bit clock usually varies depending on sample rate +whilst sync runs at the sample rate. PCM also supports Time Division +Multiplexing (TDM) in that several devices can use the bus simultaneously (this +is sometimes referred to as network mode). + +Common PCM operating modes:- + + o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. + + o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt new file mode 100644 index 000000000..0110180b7 --- /dev/null +++ b/Documentation/sound/alsa/soc/DPCM.txt @@ -0,0 +1,380 @@ +Dynamic PCM +=========== + +1. Description +============== + +Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to +various digital endpoints during the PCM stream runtime. e.g. PCM0 can route +digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP +drivers that expose several ALSA PCMs and can route to multiple DAIs. + +The DPCM runtime routing is determined by the ALSA mixer settings in the same +way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM +graph representing the DSP internal audio paths and uses the mixer settings to +determine the patch used by each ALSA PCM. + +DPCM re-uses all the existing component codec, platform and DAI drivers without +any modifications. + + +Phone Audio System with SoC based DSP +------------------------------------- + +Consider the following phone audio subsystem. This will be used in this +document for all examples :- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This diagram shows a simple smart phone audio subsystem. It supports Bluetooth, +FM digital radio, Speakers, Headset Jack, digital microphones and cellular +modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and +supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any +of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI. + + + +Example - DPCM Switching playback from DAI0 to DAI1 +--------------------------------------------------- + +Audio is being played to the Headset. After a while the user removes the headset +and audio continues playing on the speakers. + +Playback on PCM0 to Headset would look like :- + + ************* +PCM0 <============> * * <====DAI0=====> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The headset is removed from the jack by user so the speakers must now be used :- + + ************* +PCM0 <============> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The audio driver processes this as follows :- + + 1) Machine driver receives Jack removal event. + + 2) Machine driver OR audio HAL disables the Headset path. + + 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0 + for headset since the path is now disabled. + + 4) Machine driver or audio HAL enables the speaker path. + + 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and + trigger(start) for DAI1 Speakers since the path is enabled. + +In this example, the machine driver or userspace audio HAL can alter the routing +and then DPCM will take care of managing the DAI PCM operations to either bring +the link up or down. Audio playback does not stop during this transition. + + + +DPCM machine driver +=================== + +The DPCM enabled ASoC machine driver is similar to normal machine drivers +except that we also have to :- + + 1) Define the FE and BE DAI links. + + 2) Define any FE/BE PCM operations. + + 3) Define widget graph connections. + + +1 FE and BE DAI links +--------------------- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +For the example above we have to define 4 FE DAI links and 6 BE DAI links. The +FE DAI links are defined as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + { + .name = "PCM0 System", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "dsp-audio", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + .....< other FE and BE DAI links here > +}; + +This FE DAI link is pretty similar to a regular DAI link except that we also +set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream +directions should also be set with the "dpcm_playback" and "dpcm_capture" +flags. There is also an option to specify the ordering of the trigger call for +each FE. This allows the ASoC core to trigger the DSP before or after the other +components (as some DSPs have strong requirements for the ordering DAI/DSP +start and stop sequences). + +The FE DAI above sets the codec and code DAIs to dummy devices since the BE is +dynamic and will change depending on runtime config. + +The BE DAIs are configured as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + .....< FE DAI links here > + { + .name = "Codec Headset", + .cpu_dai_name = "ssp-dai.0", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "rt5640.0-001c", + .codec_dai_name = "rt5640-aif1", + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = hswult_ssp0_fixup, + .ops = &haswell_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + .....< other BE DAI links here > +}; + +This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets +the "no_pcm" flag to mark it has a BE and sets flags for supported stream +directions using "dpcm_playback" and "dpcm_capture" above. + +The BE has also flags set for ignoring suspend and PM down time. This allows +the BE to work in a hostless mode where the host CPU is not transferring data +like a BT phone call :- + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <====DAI3=====> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are +still in operation. + +A BE DAI link can also set the codec to a dummy device if the code is a device +that is managed externally. + +Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the +DSP firmware. + + +2 FE/BE PCM operations +---------------------- + +The BE above also exports some PCM operations and a "fixup" callback. The fixup +callback is used by the machine driver to (re)configure the DAI based upon the +FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE. + +e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for +DAI0. This means all FE hw_params have to be fixed in the machine driver for +DAI0 so that the DAI is running at desired configuration regardless of the FE +configuration. + +static int dai0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set DAI0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +The other PCM operation are the same as for regular DAI links. Use as necessary. + + +3 Widget graph connections +-------------------------- + +The BE DAI links will normally be connected to the graph at initialisation time +by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this +has to be set explicitly in the driver :- + +/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */ +{"DAI0 CODEC IN", NULL, "AIF1 Capture"}, +{"AIF1 Playback", NULL, "DAI0 CODEC OUT"}, + + +Writing a DPCM DSP driver +========================= + +The DPCM DSP driver looks much like a standard platform class ASoC driver +combined with elements from a codec class driver. A DSP platform driver must +implement :- + + 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver. + + 2) DAPM graph showing DSP audio routing from FE DAIs to BEs. + + 3) DAPM widgets from DSP graph. + + 4) Mixers for gains, routing, etc. + + 5) DMA configuration. + + 6) BE AIF widgets. + +Items 6 is important for routing the audio outside of the DSP. AIF need to be +defined for each BE and each stream direction. e.g for BE DAI0 above we would +have :- + +SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0), + +The BE AIF are used to connect the DSP graph to the graphs for the other +component drivers (e.g. codec graph). + + +Hostless PCM streams +==================== + +A hostless PCM stream is a stream that is not routed through the host CPU. An +example of this would be a phone call from handset to modem. + + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +In this case the PCM data is routed via the DSP. The host CPU in this use case +is only used for control and can sleep during the runtime of the stream. + +The host can control the hostless link either by :- + + 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link + is enabled or disabled by the state of the DAPM graph. This usually means + there is a mixer control that can be used to connect or disconnect the path + between both DAIs. + + 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM + graph. Control is then carried out by the FE as regular PCM operations. + This method gives more control over the DAI links, but requires much more + userspace code to control the link. Its recommended to use CODEC<->CODEC + unless your HW needs more fine grained sequencing of the PCM ops. + + +CODEC <-> CODEC link +-------------------- + +This DAI link is enabled when DAPM detects a valid path within the DAPM graph. +The machine driver sets some additional parameters to the DAI link i.e. + +static const struct snd_soc_pcm_stream dai_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link dais[] = { + < ... more DAI links above ... > + { + .name = "MODEM", + .stream_name = "MODEM", + .cpu_dai_name = "dai2", + .codec_dai_name = "modem-aif1", + .codec_name = "modem", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .params = &dai_params, + } + < ... more DAI links here ... > + +These parameters are used to configure the DAI hw_params() when DAPM detects a +valid path and then calls the PCM operations to start the link. DAPM will also +call the appropriate PCM operations to disable the DAI when the path is no +longer valid. + + +Hostless FE +----------- + +The DAI link(s) are enabled by a FE that does not read or write any PCM data. +This means creating a new FE that is connected with a virtual path to both +DAI links. The DAI links will be started when the FE PCM is started and stopped +when the FE PCM is stopped. Note that the FE PCM cannot read or write data in +this configuration. + + diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt new file mode 100644 index 000000000..b1300162e --- /dev/null +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -0,0 +1,51 @@ +Audio Clocking +============== + +This text describes the audio clocking terms in ASoC and digital audio in +general. Note: Audio clocking can be complex! + + +Master Clock +------------ + +Every audio subsystem is driven by a master clock (sometimes referred to as MCLK +or SYSCLK). This audio master clock can be derived from a number of sources +(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct +audio playback and capture sample rates. + +Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that +their speed can be altered by software (depending on the system use and to save +power). Other master clocks are fixed at a set frequency (i.e. crystals). + + +DAI Clocks +---------- +The Digital Audio Interface is usually driven by a Bit Clock (often referred to +as BCLK). This clock is used to drive the digital audio data across the link +between the codec and CPU. + +The DAI also has a frame clock to signal the start of each audio frame. This +clock is sometimes referred to as LRC (left right clock) or FRAME. This clock +runs at exactly the sample rate (LRC = Rate). + +Bit Clock can be generated as follows:- + +BCLK = MCLK / x + + or + +BCLK = LRC * x + + or + +BCLK = LRC * Channels * Word Size + +This relationship depends on the codec or SoC CPU in particular. In general +it is best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and word size) to save on power. + +It is also desirable to use the codec (if possible) to drive (or master) the +audio clocks as it usually gives more accurate sample rates than the CPU. + + + diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt new file mode 100644 index 000000000..db5f9c9ae --- /dev/null +++ b/Documentation/sound/alsa/soc/codec.txt @@ -0,0 +1,179 @@ +ASoC Codec Class Driver +======================= + +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. + +Each codec class driver *must* provide the following features:- + + 1) Codec DAI and PCM configuration + 2) Codec control IO - using RegMap API + 3) Mixers and audio controls + 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. + +Optionally, codec drivers can also provide:- + + 7) DAC Digital mute control. + +Its probably best to use this guide in conjunction with the existing codec +driver code in sound/soc/codecs/ + +ASoC Codec driver breakdown +=========================== + +1 - Codec DAI and PCM configuration +----------------------------------- +Each codec driver must have a struct snd_soc_dai_driver to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be +registered with the core by your machine driver. + +e.g. + +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + +struct snd_soc_dai_driver wm8731_dai = { + .name = "wm8731-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8731_RATES, + .formats = WM8731_FORMATS,}, + .ops = &wm8731_dai_ops, + .symmetric_rates = 1, +}; + + +2 - Codec control IO +-------------------- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. + + +3 - Mixers and audio controls +----------------------------- +All the codec mixers and audio controls can be defined using the convenience +macros defined in soc.h. + + #define SOC_SINGLE(xname, reg, shift, mask, invert) + +Defines a single control as follows:- + + xname = Control name e.g. "Playback Volume" + reg = codec register + shift = control bit(s) offset in register + mask = control bit size(s) e.g. mask of 7 = 3 bits + invert = the control is inverted + +Other macros include:- + + #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) + +A stereo control + + #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) + +A stereo control spanning 2 registers + + #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) + +Defines an single enumerated control as follows:- + + xreg = register + xshift = control bit(s) offset in register + xmask = control bit(s) size + xtexts = pointer to array of strings that describe each setting + + #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) + +Defines a stereo enumerated control + + +4 - Codec Audio Operations +-------------------------- +The codec driver also supports the following ALSA PCM operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); +}; + +Please refer to the ALSA driver PCM documentation for details. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ + + +5 - DAPM description. +--------------------- +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. + +Please also see the examples in other codec drivers. + + +6 - DAPM event handler +---------------------- +This function is a callback that handles codec domain PM calls and system +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. + +Power states:- + + SNDRV_CTL_POWER_D0: /* full On */ + /* vref/mid, clk and osc on, active */ + + SNDRV_CTL_POWER_D1: /* partial On */ + SNDRV_CTL_POWER_D2: /* partial On */ + + SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* everything off except vref/vmid, inactive */ + + SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ + + +7 - Codec DAC digital mute control +---------------------------------- +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. + +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. + +i.e. + +static int wm8974_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + else + snd_soc_write(codec, WM8974_DAC, mute_reg); + return 0; +} diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt new file mode 100644 index 000000000..6faab4880 --- /dev/null +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -0,0 +1,305 @@ +Dynamic Audio Power Management for Portable Devices +=================================================== + +1. Description +============== + +Dynamic Audio Power Management (DAPM) is designed to allow portable +Linux devices to use the minimum amount of power within the audio +subsystem at all times. It is independent of other kernel PM and as +such, can easily co-exist with the other PM systems. + +DAPM is also completely transparent to all user space applications as +all power switching is done within the ASoC core. No code changes or +recompiling are required for user space applications. DAPM makes power +switching decisions based upon any audio stream (capture/playback) +activity and audio mixer settings within the device. + +DAPM spans the whole machine. It covers power control within the entire +audio subsystem, this includes internal codec power blocks and machine +level power systems. + +There are 4 power domains within DAPM + + 1. Codec bias domain - VREF, VMID (core codec and audio power) + Usually controlled at codec probe/remove and suspend/resume, although + can be set at stream time if power is not needed for sidetone, etc. + + 2. Platform/Machine domain - physically connected inputs and outputs + Is platform/machine and user action specific, is configured by the + machine driver and responds to asynchronous events e.g when HP + are inserted + + 3. Path domain - audio subsystem signal paths + Automatically set when mixer and mux settings are changed by the user. + e.g. alsamixer, amixer. + + 4. Stream domain - DACs and ADCs. + Enabled and disabled when stream playback/capture is started and + stopped respectively. e.g. aplay, arecord. + +All DAPM power switching decisions are made automatically by consulting an audio +routing map of the whole machine. This map is specific to each machine and +consists of the interconnections between every audio component (including +internal codec components). All audio components that effect power are called +widgets hereafter. + + +2. DAPM Widgets +=============== + +Audio DAPM widgets fall into a number of types:- + + o Mixer - Mixes several analog signals into a single analog signal. + o Mux - An analog switch that outputs only one of many inputs. + o PGA - A programmable gain amplifier or attenuation widget. + o ADC - Analog to Digital Converter + o DAC - Digital to Analog Converter + o Switch - An analog switch + o Input - A codec input pin + o Output - A codec output pin + o Headphone - Headphone (and optional Jack) + o Mic - Mic (and optional Jack) + o Line - Line Input/Output (and optional Jack) + o Speaker - Speaker + o Supply - Power or clock supply widget used by other widgets. + o Regulator - External regulator that supplies power to audio components. + o Clock - External clock that supplies clock to audio components. + o AIF IN - Audio Interface Input (with TDM slot mask). + o AIF OUT - Audio Interface Output (with TDM slot mask). + o Siggen - Signal Generator. + o DAI IN - Digital Audio Interface Input. + o DAI OUT - Digital Audio Interface Output. + o DAI Link - DAI Link between two DAI structures */ + o Pre - Special PRE widget (exec before all others) + o Post - Special POST widget (exec after all others) + +(Widgets are defined in include/sound/soc-dapm.h) + +Widgets can be added to the sound card by any of the component driver types. +There are convenience macros defined in soc-dapm.h that can be used to quickly +build a list of widgets of the codecs and machines DAPM widgets. + +Most widgets have a name, register, shift and invert. Some widgets have extra +parameters for stream name and kcontrols. + + +2.1 Stream Domain Widgets +------------------------- + +Stream Widgets relate to the stream power domain and only consist of ADCs +(analog to digital converters), DACs (digital to analog converters), +AIF IN and AIF OUT. + +Stream widgets have the following format:- + +SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), +SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) + +NOTE: the stream name must match the corresponding stream name in your codec +snd_soc_codec_dai. + +e.g. stream widgets for HiFi playback and capture + +SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), +SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), + +e.g. stream widgets for AIF + +SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + +2.2 Path Domain Widgets +----------------------- + +Path domain widgets have a ability to control or affect the audio signal or +audio paths within the audio subsystem. They have the following form:- + +SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) + +Any widget kcontrols can be set using the controls and num_controls members. + +e.g. Mixer widget (the kcontrols are declared first) + +/* Output Mixer */ +static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), +}; + +SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, + ARRAY_SIZE(wm8731_output_mixer_controls)), + +If you dont want the mixer elements prefixed with the name of the mixer widget, +you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same +as for SND_SOC_DAPM_MIXER. + + +2.3 Machine domain Widgets +-------------------------- + +Machine widgets are different from codec widgets in that they don't have a +codec register bit associated with them. A machine widget is assigned to each +machine audio component (non codec or DSP) that can be independently +powered. e.g. + + o Speaker Amp + o Microphone Bias + o Jack connectors + +A machine widget can have an optional call back. + +e.g. Jack connector widget for an external Mic that enables Mic Bias +when the Mic is inserted:- + +static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) +{ + gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), + + +2.4 Codec (BIAS) Domain +----------------------- + +The codec bias power domain has no widgets and is handled by the codecs DAPM +event handler. This handler is called when the codec powerstate is changed wrt +to any stream event or by kernel PM events. + + +2.5 Virtual Widgets +------------------- + +Sometimes widgets exist in the codec or machine audio map that don't have any +corresponding soft power control. In this case it is necessary to create +a virtual widget - a widget with no control bits e.g. + +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), + +This can be used to merge to signal paths together in software. + +After all the widgets have been defined, they can then be added to the DAPM +subsystem individually with a call to snd_soc_dapm_new_control(). + + +3. Codec/DSP Widget Interconnections +==================================== + +Widgets are connected to each other within the codec, platform and machine by +audio paths (called interconnections). Each interconnection must be defined in +order to create a map of all audio paths between widgets. + +This is easiest with a diagram of the codec or DSP (and schematic of the machine +audio system), as it requires joining widgets together via their audio signal +paths. + +e.g., from the WM8731 output mixer (wm8731.c) + +The WM8731 output mixer has 3 inputs (sources) + + 1. Line Bypass Input + 2. DAC (HiFi playback) + 3. Mic Sidetone Input + +Each input in this example has a kcontrol associated with it (defined in example +above) and is connected to the output mixer via its kcontrol name. We can now +connect the destination widget (wrt audio signal) with its source widgets. + + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + +So we have :- + + Destination Widget <=== Path Name <=== Source Widget + +Or:- + + Sink, Path, Source + +Or :- + + "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch". + +When there is no path name connecting widgets (e.g. a direct connection) we +pass NULL for the path name. + +Interconnections are created with a call to:- + +snd_soc_dapm_connect_input(codec, sink, path, source); + +Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +interconnections have been registered with the core. This causes the core to +scan the codec and machine so that the internal DAPM state matches the +physical state of the machine. + + +3.1 Machine Widget Interconnections +----------------------------------- +Machine widget interconnections are created in the same way as codec ones and +directly connect the codec pins to machine level widgets. + +e.g. connects the speaker out codec pins to the internal speaker. + + /* ext speaker connected to codec pins LOUT2, ROUT2 */ + {"Ext Spk", NULL , "ROUT2"}, + {"Ext Spk", NULL , "LOUT2"}, + +This allows the DAPM to power on and off pins that are connected (and in use) +and pins that are NC respectively. + + +4 Endpoint Widgets +=================== +An endpoint is a start or end point (widget) of an audio signal within the +machine and includes the codec. e.g. + + o Headphone Jack + o Internal Speaker + o Internal Mic + o Mic Jack + o Codec Pins + +Endpoints are added to the DAPM graph so that their usage can be determined in +order to save power. e.g. NC codecs pins will be switched OFF, unconnected +jacks can also be switched OFF. + + +5 DAPM Widget Events +==================== + +Some widgets can register their interest with the DAPM core in PM events. +e.g. A Speaker with an amplifier registers a widget so the amplifier can be +powered only when the spk is in use. + +/* turn speaker amplifier on/off depending on use */ +static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) +{ + gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* corgi machine dapm widgets */ +static const struct snd_soc_dapm_widget wm8731_dapm_widgets = + SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + +Please see soc-dapm.h for all other widgets that support events. + + +5.1 Event types +--------------- + +The following event types are supported by event widgets. + +/* dapm event types */ +#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ +#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ +#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ +#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ +#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ +#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt new file mode 100644 index 000000000..fcf82a417 --- /dev/null +++ b/Documentation/sound/alsa/soc/jack.txt @@ -0,0 +1,71 @@ +ASoC jack detection +=================== + +ALSA has a standard API for representing physical jacks to user space, +the kernel side of which can be seen in include/sound/jack.h. ASoC +provides a version of this API adding two additional features: + + - It allows more than one jack detection method to work together on one + user visible jack. In embedded systems it is common for multiple + to be present on a single jack but handled by separate bits of + hardware. + + - Integration with DAPM, allowing DAPM endpoints to be updated + automatically based on the detected jack status (eg, turning off the + headphone outputs if no headphones are present). + +This is done by splitting the jacks up into three things working +together: the jack itself represented by a struct snd_soc_jack, sets of +snd_soc_jack_pins representing DAPM endpoints to update and blocks of +code providing jack reporting mechanisms. + +For example, a system may have a stereo headset jack with two reporting +mechanisms, one for the headphone and one for the microphone. Some +systems won't be able to use their speaker output while a headphone is +connected and so will want to make sure to update both speaker and +headphone when the headphone jack status changes. + +The jack - struct snd_soc_jack +============================== + +This represents a physical jack on the system and is what is visible to +user space. The jack itself is completely passive, it is set up by the +machine driver and updated by jack detection methods. + +Jacks are created by the machine driver calling snd_soc_jack_new(). + +snd_soc_jack_pin +================ + +These represent a DAPM pin to update depending on some of the status +bits supported by the jack. Each snd_soc_jack has zero or more of these +which are updated automatically. They are created by the machine driver +and associated with the jack using snd_soc_jack_add_pins(). The status +of the endpoint may configured to be the opposite of the jack status if +required (eg, enabling a built in microphone if a microphone is not +connected via a jack). + +Jack detection methods +====================== + +Actual jack detection is done by code which is able to monitor some +input to the system and update a jack by calling snd_soc_jack_report(), +specifying a subset of bits to update. The jack detection code should +be set up by the machine driver, taking configuration for the jack to +update and the set of things to report when the jack is connected. + +Often this is done based on the status of a GPIO - a handler for this is +provided by the snd_soc_jack_add_gpio() function. Other methods are +also available, for example integrated into CODECs. One example of +CODEC integrated jack detection can be see in the WM8350 driver. + +Each jack may have multiple reporting mechanisms, though it will need at +least one to be useful. + +Machine drivers +=============== + +These are all hooked together by the machine driver depending on the +system hardware. The machine driver will set up the snd_soc_jack and +the list of pins to update then set up one or more jack detection +mechanisms to update that jack based on their current status. diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt new file mode 100644 index 000000000..74056dba5 --- /dev/null +++ b/Documentation/sound/alsa/soc/machine.txt @@ -0,0 +1,93 @@ +ASoC Machine Driver +=================== + +The ASoC machine (or board) driver is the code that glues together all the +component drivers (e.g. codecs, platforms and DAIs). It also describes the +relationships between each componnent which include audio paths, GPIOs, +interrupts, clocking, jacks and voltage regulators. + +The machine driver can contain codec and platform specific code. It registers +the audio subsystem with the kernel as a platform device and is represented by +the following struct:- + +/* SoC machine */ +struct snd_soc_card { + char *name; + + ... + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + + /* the pre and post PM functions are used to do any PM work before and + * after the codec and DAIs do any PM work. */ + int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); + int (*suspend_post)(struct platform_device *pdev, pm_message_t state); + int (*resume_pre)(struct platform_device *pdev); + int (*resume_post)(struct platform_device *pdev); + + ... + + /* CPU <--> Codec DAI links */ + struct snd_soc_dai_link *dai_link; + int num_links; + + ... +}; + +probe()/remove() +---------------- +probe/remove are optional. Do any machine specific probe here. + + +suspend()/resume() +------------------ +The machine driver has pre and post versions of suspend and resume to take care +of any machine audio tasks that have to be done before or after the codec, DAIs +and DMA is suspended and resumed. Optional. + + +Machine DAI Configuration +------------------------- +The machine DAI configuration glues all the codec and CPU DAIs together. It can +also be used to set up the DAI system clock and for any machine related DAI +initialisation e.g. the machine audio map can be connected to the codec audio +map, unconnected codec pins can be set as such. + +struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. + +/* corgi digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link corgi_dai = { + .name = "WM8731", + .stream_name = "WM8731", + .cpu_dai_name = "pxa-is2-dai", + .codec_dai_name = "wm8731-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "wm8713-codec.0-001a", + .init = corgi_wm8731_init, + .ops = &corgi_ops, +}; + +struct snd_soc_card then sets up the machine with its DAIs. e.g. + +/* corgi audio machine driver */ +static struct snd_soc_card snd_soc_corgi = { + .name = "Corgi", + .dai_link = &corgi_dai, + .num_links = 1, +}; + + +Machine Power Map +----------------- + +The machine driver can optionally extend the codec power map and to become an +audio power map of the audio subsystem. This allows for automatic power up/down +of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack +sockets in the machine init function. + + +Machine Controls +---------------- + +Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt new file mode 100644 index 000000000..ff88f52ee --- /dev/null +++ b/Documentation/sound/alsa/soc/overview.txt @@ -0,0 +1,95 @@ +ALSA SoC Layer +============== + +The overall project goal of the ALSA System on Chip (ASoC) layer is to +provide better ALSA support for embedded system-on-chip processors (e.g. +pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC +subsystem there was some support in the kernel for SoC audio, however it +had some limitations:- + + * Codec drivers were often tightly coupled to the underlying SoC + CPU. This is not ideal and leads to code duplication - for example, + Linux had different wm8731 drivers for 4 different SoC platforms. + + * There was no standard method to signal user initiated audio events (e.g. + Headphone/Mic insertion, Headphone/Mic detection after an insertion + event). These are quite common events on portable devices and often require + machine specific code to re-route audio, enable amps, etc., after such an + event. + + * Drivers tended to power up the entire codec when playing (or + recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There was also no support for saving + power via changing codec oversampling rates, bias currents, etc. + + +ASoC Design +=========== + +The ASoC layer is designed to address these issues and provide the following +features :- + + * Codec independence. Allows reuse of codec drivers on other platforms + and machines. + + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC + interface and codec registers its audio interface capabilities with the + core and are subsequently matched and configured when the application + hardware parameters are known. + + * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to + its minimum power state at all times. This includes powering up/down + internal power blocks depending on the internal codec audio routing and any + active streams. + + * Pop and click reduction. Pops and clicks can be reduced by powering the + codec up/down in the correct sequence (including using digital mute). ASoC + signals the codec when to change power states. + + * Machine specific controls: Allow machines to add controls to the sound card + (e.g. volume control for speaker amplifier). + +To achieve all this, ASoC basically splits an embedded audio system into +multiple re-usable component drivers :- + + * Codec class drivers: The codec class driver is platform independent and + contains audio controls, audio interface capabilities, codec DAPM + definition and codec IO functions. This class extends to BT, FM and MODEM + ICs if required. Codec class drivers should be generic code that can run + on any architecture and machine. + + * Platform class drivers: The platform class driver includes the audio DMA + engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM) + and any audio DSP drivers for that platform. + + * Machine class driver: The machine driver class acts as the glue that + decribes and binds the other component drivers together to form an ALSA + "sound card device". It handles any machine specific controls and + machine level audio events (e.g. turning on an amp at start of playback). + + +Documentation +============= + +The documentation is spilt into the following sections:- + +overview.txt: This file. + +codec.txt: Codec driver internals. + +DAI.txt: Description of Digital Audio Interface standards and how to configure +a DAI within your codec and CPU DAI drivers. + +dapm.txt: Dynamic Audio Power Management + +platform.txt: Platform audio DMA and DAI. + +machine.txt: Machine driver internals. + +pop_clicks.txt: How to minimise audio artifacts. + +clocking.txt: ASoC clocking for best power performance. + +jack.txt: ASoC jack detection. + +DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt new file mode 100644 index 000000000..3a08a2c91 --- /dev/null +++ b/Documentation/sound/alsa/soc/platform.txt @@ -0,0 +1,79 @@ +ASoC Platform Driver +==================== + +An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI +drivers and DSP drivers. The platform drivers only target the SoC CPU and must +have no board specific code. + +Audio DMA +========= + +The platform DMA driver optionally supports the following ALSA operations:- + +/* SoC audio ops */ +struct snd_soc_ops { + int (*startup)(struct snd_pcm_substream *); + void (*shutdown)(struct snd_pcm_substream *); + int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); + int (*hw_free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); + int (*trigger)(struct snd_pcm_substream *, int); +}; + +The platform driver exports its DMA functionality via struct +snd_soc_platform_driver:- + +struct snd_soc_platform_driver { + char *name; + + int (*probe)(struct platform_device *pdev); + int (*remove)(struct platform_device *pdev); + int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + + /* pcm creation and destruction */ + int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + void (*pcm_free)(struct snd_pcm *); + + /* + * For platform caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); + + /* platform stream ops */ + struct snd_pcm_ops *pcm_ops; +}; + +Please refer to the ALSA driver documentation for details of audio DMA. +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ + +An example DMA driver is soc/pxa/pxa2xx-pcm.c + + +SoC DAI Drivers +=============== + +Each SoC DAI driver must provide the following features:- + + 1) Digital audio interface (DAI) description + 2) Digital audio interface configuration + 3) PCM's description + 4) SYSCLK configuration + 5) Suspend and resume (optional) + +Please see codec.txt for a description of items 1 - 4. + + +SoC DSP Drivers +=============== + +Each SoC DSP driver usually supplies the following features :- + + 1) DAPM graph + 2) Mixer controls + 3) DMA IO to/from DSP buffers (if applicable) + 4) Definition of DSP front end (FE) PCM devices. + +Please see DPCM.txt for a description of item 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt new file mode 100644 index 000000000..e1e74daa4 --- /dev/null +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -0,0 +1,52 @@ +Audio Pops and Clicks +===================== + +Pops and clicks are unwanted audio artifacts caused by the powering up and down +of components within the audio subsystem. This is noticeable on PCs when an +audio module is either loaded or unloaded (at module load time the sound card is +powered up and causes a popping noise on the speakers). + +Pops and clicks can be more frequent on portable systems with DAPM. This is +because the components within the subsystem are being dynamically powered +depending on the audio usage and this can subsequently cause a small pop or +click every time a component power state is changed. + + +Minimising Playback Pops and Clicks +=================================== + +Playback pops in portable audio subsystems cannot be completely eliminated +currently, however future audio codec hardware will have better pop and click +suppression. Pops can be reduced within playback by powering the audio +components in a specific order. This order is different for startup and +shutdown and follows some basic rules:- + + Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute + + Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC + +This assumes that the codec PCM output path from the DAC is via a mixer and then +a PGA (programmable gain amplifier) before being output to the speakers. + + +Minimising Capture Pops and Clicks +================================== + +Capture artifacts are somewhat easier to get rid as we can delay activating the +ADC until all the pops have occurred. This follows similar power rules to +playback in that components are powered in a sequence depending upon stream +startup or shutdown. + + Startup Order - Input PGA --> Mixers --> ADC + + Shutdown Order - ADC --> Mixers --> Input PGA + + +Zipper Noise +============ +An unwanted zipper noise can occur within the audio playback or capture stream +when a volume control is changed near its maximum gain value. The zipper noise +is heard when the gain increase or decrease changes the mean audio signal +amplitude too quickly. It can be minimised by enabling the zero cross setting +for each volume control. The ZC forces the gain change to occur when the signal +crosses the zero amplitude line. diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 000000000..0b191a23f --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->|<queued samples>|<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 |