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Diffstat (limited to 'Documentation/sound/alsa')
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diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt deleted file mode 100644 index fc53ccd9a629..000000000000 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ /dev/null @@ -1,2330 +0,0 @@ - - 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 "turtlebeach/msndinit.bin" and - "turtlebeach/msndperm.bin" 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 "turtlebeach/pndspini.bin" and - "turtlebeach/pndsperm.bin" 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 "riptide.hex" to the standard - firmware path (e.g. /lib/firmware). - - Module snd-rme32 - ---------------- - - Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, - Prodif96 and Prodif Gold) sound cards. - - This module supports multiple cards. - - Module snd-rme96 - ---------------- - - Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. - - This module supports multiple cards. - - Module snd-rme9652 - ------------------ - - Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. - - precise_ptr - Enable precise pointer (doesn't work reliably). - (default = 0) - - This module supports multiple cards. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any RME9652 cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-sa11xx-uda1341 (on arm only) - --------------------------------------- - - Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. - - Module supports only one card. - Module has no enable and index options. - - The power-management is supported. - - Module snd-sb8 - -------------- - - Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, - SoundBlaster 2.0, - SoundBlaster Pro - - port - port # for SB DSP chip (0x220,0x240,0x260) - irq - IRQ # for SB DSP chip (5,7,9,10) - dma8 - DMA # for SB DSP chip (1,3) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-sb16 and snd-sbawe - ----------------------------- - - Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), - SoundBlaster AWE 32 (PnP), - SoundBlaster AWE 64 PnP - - mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) - csp - ASP/CSP chip support - 0 = disable (default), 1 = enable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for SB DSP 4.x chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) - (snd-sbawe module only) - irq - IRQ # for SB DSP 4.x chip (5,7,9,10) - dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) - dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) - - This module supports multiple cards, autoprobe and ISA PnP. - - Note: To use Vibra16X cards in 16-bit half duplex mode, you must - disable 16bit DMA with dma16 = -1 module parameter. - Also, all Sound Blaster 16 type cards can operate in 16-bit - half duplex mode through 8-bit DMA channel by disabling their - 16-bit DMA channel. - - The power-management is supported. - - Module snd-sc6000 - ----------------- - - Module for Gallant SC-6000 soundcard 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) - ospath - Pathname to processed ICS2115 OS firmware - (default:wavefront.os) - The path name of the ISC2115 OS firmware. In the recent - version, it's handled via firmware loader framework, so it - must be installed in the proper path, typically, - /lib/firmware. - reset_time - How long to wait for a reset to take effect - (default:2) - ramcheck_time - How many seconds to wait for the RAM test - (default:20) - osrun_time - How many seconds to wait for the ICS2115 OS - (default:10) - - This module supports multiple cards and ISA PnP. - - Note: the firmware file "wavefront.os" was located in the earlier - version in /etc. Now it's loaded via firmware loader, and - must be in the proper firmware path, such as /lib/firmware. - Copy (or symlink) the file appropriately if you get an error - regarding firmware downloading after upgrading the kernel. - - Module snd-sonicvibes - --------------------- - - Module for S3 SonicVibes PCI sound cards. - * PINE Schubert 32 PCI - - reverb - Reverb Enable - 1 = enable, 0 = disable (default) - - SoundCard must have onboard SRAM for this. - mge - Mic Gain Enable - 1 = enable, 0 = disable (default) - - This module supports multiple cards and autoprobe. - - Module snd-serial-u16550 - ------------------------ - - Module for UART16550A serial MIDI ports. - - port - port # for UART16550A chip - irq - IRQ # for UART16550A chip, -1 = poll mode - speed - speed in bauds (9600,19200,38400,57600,115200) - 38400 = default - base - base for divisor in bauds (57600,115200,230400,460800) - 115200 = default - outs - number of MIDI ports in a serial port (1-4) - 1 = default - adaptor - Type of adaptor. - 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, - 3 = MS-124W M/B, 4 = Generic - - This module supports multiple cards. This module does not support autoprobe - thus the main port must be specified!!! Other options are optional. - - Module snd-trident - ------------------ - - Module for Trident 4DWave DX/NX sound cards. - * Best Union Miss Melody 4DWave PCI - * HIS 4DWave PCI - * Warpspeed ONSpeed 4DWave PCI - * AzTech PCI 64-Q3D - * Addonics SV 750 - * CHIC True Sound 4Dwave - * Shark Predator4D-PCI - * Jaton SonicWave 4D - * SiS SI7018 PCI Audio - * Hoontech SoundTrack Digital 4DWave NX - - pcm_channels - max channels (voices) reserved for PCM - wavetable_size - max wavetable size in kB (4-?kb) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-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) - autoclock - Enable auto-clock selection for UAC2 devices - (default: yes) - quirk_alias - Quirk alias list, pass strings like - "0123abcd:5678beef", which applies the existing - quirk for the device 5678:beef to a new device - 0123:abcd. - - 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. - NB: quirk_alias option is provided only for testing / development. - If you want to have a proper support, contact to upstream for - adding the matching quirk in the driver code statically. - - 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 deleted file mode 100644 index 7f10dc6ff28c..000000000000 --- a/Documentation/sound/alsa/Audigy-mixer.txt +++ /dev/null @@ -1,345 +0,0 @@ - - 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 deleted file mode 100644 index e7a5ed4dcae8..000000000000 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ /dev/null @@ -1,442 +0,0 @@ - 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 deleted file mode 100644 index f158cde8b065..000000000000 --- a/Documentation/sound/alsa/Bt87x.txt +++ /dev/null @@ -1,78 +0,0 @@ -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 deleted file mode 100644 index 4e36e6e809ca..000000000000 --- a/Documentation/sound/alsa/CMIPCI.txt +++ /dev/null @@ -1,254 +0,0 @@ - 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 deleted file mode 100644 index 3c43d1a4ca0e..000000000000 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ /dev/null @@ -1,153 +0,0 @@ -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 deleted file mode 100644 index 3fc1cf50d28e..000000000000 --- a/Documentation/sound/alsa/ControlNames.txt +++ /dev/null @@ -1,107 +0,0 @@ -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 deleted file mode 100644 index e9621e349e17..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Controls.txt +++ /dev/null @@ -1,116 +0,0 @@ -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-DP-MST-audio.txt b/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt deleted file mode 100644 index 82744ac3513d..000000000000 --- a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt +++ /dev/null @@ -1,74 +0,0 @@ -To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin -and dynamic pcm assignment. - -Virtual pin is an extension of per_pin. The most difference of DP MST -from legacy is that DP MST introduces device entry. Each pin can contain -several device entries. Each device entry behaves as a pin. - -As each pin may contain several device entries and each codec may contain -several pins, if we use one pcm per per_pin, there will be many PCMs. -The new solution is to create a few PCMs and to dynamically bind pcm to -per_pin. Driver uses spec->dyn_pcm_assign flag to indicate whether to use -the new solution. - -PCM -=== -To be added - - -Jack -==== - -Presume: - - MST must be dyn_pcm_assign, and it is acomp (for Intel scenario); - - NON-MST may or may not be dyn_pcm_assign, it can be acomp or !acomp; - -So there are the following scenarios: - a. MST (&& dyn_pcm_assign && acomp) - b. NON-MST && dyn_pcm_assign && acomp - c. NON-MST && !dyn_pcm_assign && !acomp - -Below discussion will ignore MST and NON-MST difference as it doesn't -impact on jack handling too much. - -Driver uses struct hdmi_pcm pcm[] array in hdmi_spec and snd_jack is -a member of hdmi_pcm. Each pin has one struct hdmi_pcm * pcm pointer. - -For !dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] statically. - -For dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] -when monitor is hotplugged. - - -Build Jack ----------- - -- dyn_pcm_assign -Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly. - -- !dyn_pcm_assign -Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically. - - -Unsolicited Event Enabling --------------------------- -Enable unsolicited event if !acomp. - - -Monitor Hotplug Event Handling ------------------------------- -- acomp -pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() -> -sync_eld_via_acomp(). -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for -both dyn_pcm_assign and !dyn_pcm_assign - -- !acomp -Hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() -> -hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs() -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign. -Use hda_jack mechanism to handle jack events. - - -Others to be added later -======================== diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt deleted file mode 100644 index ec099d4343f2..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ /dev/null @@ -1,324 +0,0 @@ - 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 -====== - gpio1 Enable GPIO1 - coef Enable EAPD via COEF table - fujitsu Quirk for FSC S7020 - fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support - -ALC262 -====== - inv-dmic Inverted internal mic workaround - -ALC267/268 -========== - inv-dmic Inverted internal mic workaround - hp-eapd Disable HP EAPD on NID 0x15 - -ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) -====== - 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 - headset-mode More comprehensive headset support for ALC269 & co - headset-mode-no-hp-mic Headset mode support without headphone mic - lenovo-dock Enables docking station I/O for some Lenovos - hp-gpio-led GPIO LED support on HP laptops - 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 - alc283-dac-wcaps Fixups for Chromebook with ALC283 - alc283-sense-combo Combo jack sensing on ALC283 - tpt440-dock Pin configs for Lenovo Thinkpad Dock support - -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 deleted file mode 100644 index d4510ebf2e8c..000000000000 --- a/Documentation/sound/alsa/HD-Audio.txt +++ /dev/null @@ -1,853 +0,0 @@ -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. - - -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. It's included in alsa-utils package. The latest -version can be found on git repository: - -- git://git.alsa-project.org/alsa-utils.git - -The script can be fetched directly from the following URL, too: - -- 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 program is found in the git repository below: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git - -See README file in the repository 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/Jack-Controls.txt b/Documentation/sound/alsa/Jack-Controls.txt deleted file mode 100644 index fe1c5e0c8555..000000000000 --- a/Documentation/sound/alsa/Jack-Controls.txt +++ /dev/null @@ -1,43 +0,0 @@ -Why we need Jack kcontrols -========================== - -ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...) -to user space. This means userspace applications like pulseaudio can -switch off headphones and switch on speakers when no headphones are -pluged in. - -The old ALSA jack code only created input devices for each registered -jack. These jack input devices are not readable by userspace devices -that run as non root. - -The new jack code creates embedded jack kcontrols for each jack that -can be read by any process. - -This can be combined with UCM to allow userspace to route audio more -intelligently based on jack insertion or removal events. - -Jack Kcontrol Internals -======================= - -Each jack will have a kcontrol list, so that we can create a kcontrol -and attach it to the jack, at jack creation stage. We can also add a -kcontrol to an existing jack, at anytime when required. - -Those kcontrols will be freed automatically when the Jack is freed. - -How to use jack kcontrols -========================= - -In order to keep compatibility, snd_jack_new() has been modified by -adding two params :- - - - @initial_kctl: if true, create a kcontrol and add it to the jack - list. - - @phantom_jack: Don't create a input device for phantom jacks. - -HDA jacks can set phantom_jack to true in order to create a phantom -jack and set initial_kctl to true to create an initial kcontrol with -the correct id. - -ASoC jacks should set initial_kctl as false. The pin name will be -assigned as the jack kcontrol name. diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt deleted file mode 100644 index ccda41b10f8a..000000000000 --- a/Documentation/sound/alsa/Joystick.txt +++ /dev/null @@ -1,86 +0,0 @@ -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 deleted file mode 100644 index 4ee35b4fbe4a..000000000000 --- a/Documentation/sound/alsa/MIXART.txt +++ /dev/null @@ -1,100 +0,0 @@ - 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 deleted file mode 100644 index 152ca2a3f1bd..000000000000 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ /dev/null @@ -1,305 +0,0 @@ - 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 deleted file mode 100644 index 7f8a0d325905..000000000000 --- a/Documentation/sound/alsa/Procfile.txt +++ /dev/null @@ -1,234 +0,0 @@ - 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 deleted file mode 100644 index 67b2ea1cc31d..000000000000 --- a/Documentation/sound/alsa/README.maya44 +++ /dev/null @@ -1,163 +0,0 @@ -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 deleted file mode 100644 index f4b5988f450c..000000000000 --- a/Documentation/sound/alsa/SB-Live-mixer.txt +++ /dev/null @@ -1,356 +0,0 @@ - - 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 deleted file mode 100644 index 1b0ac06ba95d..000000000000 --- a/Documentation/sound/alsa/VIA82xx-mixer.txt +++ /dev/null @@ -1,8 +0,0 @@ - - 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 deleted file mode 100644 index 0fa40679b080..000000000000 --- a/Documentation/sound/alsa/alsa-parameters.txt +++ /dev/null @@ -1,135 +0,0 @@ - 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 deleted file mode 100644 index 8ba556a131c3..000000000000 --- a/Documentation/sound/alsa/compress_offload.txt +++ /dev/null @@ -1,234 +0,0 @@ - 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 don't 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, -- Indicate 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 deleted file mode 100644 index 751d45036a05..000000000000 --- a/Documentation/sound/alsa/emu10k1-jack.txt +++ /dev/null @@ -1,74 +0,0 @@ -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/hdspm.txt b/Documentation/sound/alsa/hdspm.txt deleted file mode 100644 index 7ba31948dea7..000000000000 --- a/Documentation/sound/alsa/hdspm.txt +++ /dev/null @@ -1,362 +0,0 @@ -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/img,spdif-in.txt b/Documentation/sound/alsa/img,spdif-in.txt deleted file mode 100644 index 8b7505785fa6..000000000000 --- a/Documentation/sound/alsa/img,spdif-in.txt +++ /dev/null @@ -1,49 +0,0 @@ -The Imagination Technologies SPDIF Input controller contains the following -controls: - -name='IEC958 Capture Mask',index=0 - -This control returns a mask that shows which of the IEC958 status bits -can be read using the 'IEC958 Capture Default' control. - -name='IEC958 Capture Default',index=0 - -This control returns the status bits contained within the SPDIF stream that -is being received. The 'IEC958 Capture Mask' shows which bits can be read -from this control. - -name='SPDIF In Multi Frequency Acquire',index=0 -name='SPDIF In Multi Frequency Acquire',index=1 -name='SPDIF In Multi Frequency Acquire',index=2 -name='SPDIF In Multi Frequency Acquire',index=3 - -This control is used to attempt acquisition of up to four different sample -rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency' -control. - -When the value of this control is set to {0,0,0,0}, the rate given to hw_params -will determine the single rate the block will capture. Else, the rate given to -hw_params will be ignored, and the block will attempt capture for each of the -four sample rates set here. - -If less than four rates are required, the same rate can be specified more than -once - -name='SPDIF In Lock Frequency',index=0 - -This control returns the active capture rate, or 0 if a lock has not been -acquired - -name='SPDIF In Lock TRK',index=0 - -This control is used to modify the locking/jitter rejection characteristics -of the block. Larger values increase the locking range, but reduce jitter -rejection. - -name='SPDIF In Lock Acquire Threshold',index=0 - -This control is used to change the threshold at which a lock is acquired. - -name='SPDIF In Lock Release Threshold',index=0 - -This control is used to change the threshold at which a lock is released. diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt deleted file mode 100644 index 9657e8099228..000000000000 --- a/Documentation/sound/alsa/powersave.txt +++ /dev/null @@ -1,41 +0,0 @@ -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 deleted file mode 100644 index 9663b45f6fde..000000000000 --- a/Documentation/sound/alsa/seq_oss.html +++ /dev/null @@ -1,409 +0,0 @@ -<!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 deleted file mode 100644 index c1919559d509..000000000000 --- a/Documentation/sound/alsa/serial-u16550.txt +++ /dev/null @@ -1,88 +0,0 @@ - - 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 deleted file mode 100644 index c9679264c559..000000000000 --- a/Documentation/sound/alsa/soc/DAI.txt +++ /dev/null @@ -1,56 +0,0 @@ -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 deleted file mode 100644 index 0110180b7ac6..000000000000 --- a/Documentation/sound/alsa/soc/DPCM.txt +++ /dev/null @@ -1,380 +0,0 @@ -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 deleted file mode 100644 index b1300162e01c..000000000000 --- a/Documentation/sound/alsa/soc/clocking.txt +++ /dev/null @@ -1,51 +0,0 @@ -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 deleted file mode 100644 index db5f9c9ae149..000000000000 --- a/Documentation/sound/alsa/soc/codec.txt +++ /dev/null @@ -1,179 +0,0 @@ -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 deleted file mode 100644 index c45bd79f291e..000000000000 --- a/Documentation/sound/alsa/soc/dapm.txt +++ /dev/null @@ -1,305 +0,0 @@ -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 don't 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 deleted file mode 100644 index fcf82a417293..000000000000 --- a/Documentation/sound/alsa/soc/jack.txt +++ /dev/null @@ -1,71 +0,0 @@ -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 deleted file mode 100644 index 6bf2d2063b52..000000000000 --- a/Documentation/sound/alsa/soc/machine.txt +++ /dev/null @@ -1,93 +0,0 @@ -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 component 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 deleted file mode 100644 index f3f28b7ae242..000000000000 --- a/Documentation/sound/alsa/soc/overview.txt +++ /dev/null @@ -1,95 +0,0 @@ -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 - describes 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 deleted file mode 100644 index 3a08a2c9150c..000000000000 --- a/Documentation/sound/alsa/soc/platform.txt +++ /dev/null @@ -1,79 +0,0 @@ -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 deleted file mode 100644 index e1e74daa4497..000000000000 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ /dev/null @@ -1,52 +0,0 @@ -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 deleted file mode 100644 index 9d579aefbffd..000000000000 --- a/Documentation/sound/alsa/timestamping.txt +++ /dev/null @@ -1,200 +0,0 @@ -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 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 nature of DMA transfers. - -The app time corresponds to the time tracked by an application after -writing in the ring buffer. - -The application can query the hardware capabilities, define which -audio time it wants reported by selecting the relevant settings in -audio_tstamp_config fields, thus 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 subsystem, 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 but -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 |