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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 18:49:45 +0000 |
| commit | 2c3c1048746a4622d8c89a29670120dc8fab93c4 (patch) | |
| tree | 848558de17fb3008cdf4d861b01ac7781903ce39 /Documentation/sound | |
| parent | Initial commit. (diff) | |
| download | linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.tar.xz linux-2c3c1048746a4622d8c89a29670120dc8fab93c4.zip | |
Adding upstream version 6.1.76.upstream/6.1.76upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'Documentation/sound')
50 files changed, 16917 insertions, 0 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst new file mode 100644 index 000000000..21ab5e6f7 --- /dev/null +++ b/Documentation/sound/alsa-configuration.rst @@ -0,0 +1,2752 @@ +============================================================== +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. + + +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. + + +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 + +Module snd-soc-core +------------------- + +The soc core module. It is used by all ALSA card drivers. +It takes the following options which have global effects. + +prealloc_buffer_size_kbytes + Specify prealloc buffer size in kbytes (default: 512). + +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 +https://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, + 1 = integrated PCI port (default on 8738), + 0 = disable +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 + 5 = SKL+: apply the delay calculation available on recent Intel chips + 6 = FIFO: correct the position with the fixed FIFO size, for recent AMD chips +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/notes.rst 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 hd-audio/notes.rst 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.rst. + +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. + +The model option can be used also for aliasing to another PCI or codec +SSID. When it's passed in the form of ``model=XXXX:YYYY`` where XXXX +and YYYY are the sub-vendor and sub-device IDs in hex numbers, +respectively, the driver will refer to that SSID as a reference to the +quirk table. + +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.rst 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.rst 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 siree). + +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 Documentation/sound/cards/mixart.rst 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/cards/multisound.sh 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 +https://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. +implicit_fb + Apply the generic implicit feedback sync mode. When this is set + and the playback stream sync mode is ASYNC, the driver tries to + tie an adjacent ASYNC capture stream as the implicit feedback + source. This is equivalent with quirk_flags bit 17. +use_vmalloc + Use vmalloc() for allocations of the PCM buffers (default: yes). + For architectures with non-coherent memory like ARM or MIPS, the + mmap access may give inconsistent results with vmalloc'ed + buffers. If mmap is used on such architectures, turn off this + option, so that the DMA-coherent buffers are allocated and used + instead. +delayed_register + The option is needed for devices that have multiple streams + defined in multiple USB interfaces. The driver may invoke + registrations multiple times (once per interface) and this may + lead to the insufficient device enumeration. + This option receives an array of strings, and you can pass + ID:INTERFACE like ``0123abcd:4`` for performing the delayed + registration to the given device. In this example, when a USB + device 0123:abcd is probed, the driver waits the registration + until the USB interface 4 gets probed. + The driver prints a message like "Found post-registration device + assignment: 1234abcd:04" for such a device, so that user can + notice the need. +quirk_flags + Contains the bit flags for various device specific workarounds. + Applied to the corresponding card index. + + * bit 0: Skip reading sample rate for devices + * bit 1: Create Media Controller API entries + * bit 2: Allow alignment on audio sub-slot at transfer + * bit 3: Add length specifier to transfers + * bit 4: Start playback stream at first in implement feedback mode + * bit 5: Skip clock selector setup + * bit 6: Ignore errors from clock source search + * bit 7: Indicates ITF-USB DSD based DACs + * bit 8: Add a delay of 20ms at each control message handling + * bit 9: Add a delay of 1-2ms at each control message handling + * bit 10: Add a delay of 5-6ms at each control message handling + * bit 11: Add a delay of 50ms at each interface setup + * bit 12: Perform sample rate validations at probe + * bit 13: Disable runtime PM autosuspend + * bit 14: Ignore errors for mixer access + * bit 15: Support generic DSD raw U32_BE format + * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode + +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. This workaround corresponds to +the ``quirk_flags`` bit 14, too. + +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. +Ditto for ``quirk_flags``. If a device is known to require specific +workarounds, please report to the upstream. + +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 +------------------------------- +erase + erase all additional information about OSS applications + +<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 + https://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst new file mode 100644 index 000000000..f3f4640ee --- /dev/null +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -0,0 +1,368 @@ +============================================= +Sound Blaster Audigy mixer / default DSP code +============================================= + +This is based on sb-live-mixer.rst. + +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: + + +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. + + +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) + + + +MANUALS/PATENTS +=============== + +ftp://opensource.creative.com/pub/doc +------------------------------------- + +LM4545.pdf + AC97 Codec + +m2049.pdf + The EMU10K1 Digital Audio Processor + +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + +WO 9901813 (A1) + Audio Effects Processor with multiple asynchronous streams + (Jan. 14, 1999) + +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 (https://www.uspto.gov/) +----------------------------------- + +US 5925841 + Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + +US 5928342 + Audio Effects Processor integrated on a single chip + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + (Jul. 27, 1999) + +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 + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + (Oct. 24, 2000) + +US 6151670 + Method for conserving memory storage using a + pool of short term memory registers + (Nov. 21, 2000) + +US 6195715 + Interrupt control for multiple programs communicating with + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register + (Feb. 27, 2001) diff --git a/Documentation/sound/cards/audiophile-usb.rst b/Documentation/sound/cards/audiophile-usb.rst new file mode 100644 index 000000000..a7bb56483 --- /dev/null +++ b/Documentation/sound/cards/audiophile-usb.rst @@ -0,0 +1,550 @@ +======================================================== +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 + + +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) + + +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. + + +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``. + +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. + +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) + +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. + +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 + + +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) + +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 + +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. + +Technical details for hackers +----------------------------- + +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +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 + +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. + +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 + +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 + +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. + +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/cards/bt87x.rst b/Documentation/sound/cards/bt87x.rst new file mode 100644 index 000000000..912732d3e --- /dev/null +++ b/Documentation/sound/cards/bt87x.rst @@ -0,0 +1,83 @@ +================= +ALSA BT87x Driver +================= + +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/cards/cmipci.rst b/Documentation/sound/cards/cmipci.rst new file mode 100644 index 000000000..9ea1de6ec --- /dev/null +++ b/Documentation/sound/cards/cmipci.rst @@ -0,0 +1,272 @@ +================================================= +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. + + +.. [#] + 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/cards/emu10k1-jack.rst b/Documentation/sound/cards/emu10k1-jack.rst new file mode 100644 index 000000000..6597f1ea8 --- /dev/null +++ b/Documentation/sound/cards/emu10k1-jack.rst @@ -0,0 +1,78 @@ +================================================================= +Low latency, multichannel audio with JACK and the emu10k1/emu10k2 +================================================================= + +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 + + +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.rst (or audigy-mixer.rst). + +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/cards/hdspm.rst b/Documentation/sound/cards/hdspm.rst new file mode 100644 index 000000000..5373e51ed --- /dev/null +++ b/Documentation/sound/cards/hdspm.rst @@ -0,0 +1,379 @@ +======================================= +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:: + (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:: + 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:: + (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:: + 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" + +.. note:: + !!!! 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" + +.. warning:: + !!!! 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:: + 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:: + 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:: + 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:: + 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 :: + note: here the analog output is enabled (but not routed). diff --git a/Documentation/sound/cards/img-spdif-in.rst b/Documentation/sound/cards/img-spdif-in.rst new file mode 100644 index 000000000..7df9f5ae2 --- /dev/null +++ b/Documentation/sound/cards/img-spdif-in.rst @@ -0,0 +1,53 @@ +================================================ +Imagination Technologies SPDIF Input Controllers +================================================ + +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/cards/index.rst b/Documentation/sound/cards/index.rst new file mode 100644 index 000000000..c016f8c3b --- /dev/null +++ b/Documentation/sound/cards/index.rst @@ -0,0 +1,19 @@ +Card-Specific Information +========================= + +.. toctree:: + :maxdepth: 2 + + joystick + cmipci + sb-live-mixer + audigy-mixer + emu10k1-jack + via82xx-mixer + audiophile-usb + mixart + bt87x + maya44 + hdspm + serial-u16550 + img-spdif-in diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst new file mode 100644 index 000000000..488946fc1 --- /dev/null +++ b/Documentation/sound/cards/joystick.rst @@ -0,0 +1,91 @@ +======================================= +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/joydev/joystick.rst. + +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 [#f1]_ joystick 0 = disable (default), 1 = enable +ymfpci joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x201, 0x202, 0x204, 0x205 [#f2]_ +============== ============= ============================================ + +.. [#f1] VIA686A/B only +.. [#f2] 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/cards/maya44.rst b/Documentation/sound/cards/maya44.rst new file mode 100644 index 000000000..bf09a584b --- /dev/null +++ b/Documentation/sound/cards/maya44.rst @@ -0,0 +1,186 @@ +================================= +Notes on Maya44 USB Audio Support +================================= + +.. 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 + +Feb 14, 2008 + +Rainer Zimmermann <mail@lightshed.de> + +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 + diff --git a/Documentation/sound/cards/mixart.rst b/Documentation/sound/cards/mixart.rst new file mode 100644 index 000000000..48aba98b0 --- /dev/null +++ b/Documentation/sound/cards/mixart.rst @@ -0,0 +1,110 @@ +============================================================== +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/cards/multisound.sh b/Documentation/sound/cards/multisound.sh new file mode 100755 index 000000000..a915a1aff --- /dev/null +++ b/Documentation/sound/cards/multisound.sh @@ -0,0 +1,1139 @@ +#! /bin/sh +# +# Turtle Beach MultiSound Driver Notes +# -- Andrew Veliath <andrewtv@usa.net> +# +# Last update: September 10, 1998 +# Corresponding msnd driver: 0.8.3 +# +# ** This file is a README (top part) and shell archive (bottom part). +# The corresponding archived utility sources can be unpacked by +# running `sh MultiSound' (the utilities are only needed for the +# Pinnacle and Fiji cards). ** +# +# +# -=-=- Getting Firmware -=-=- +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# See the section `Obtaining and Creating Firmware Files' in this +# document for instructions on obtaining the necessary firmware +# files. +# +# +# Supported Features +# ~~~~~~~~~~~~~~~~~~ +# +# Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is +# not currently available) and mixer functionality (/dev/mixer) are +# supported (memory mapped digital audio is not yet supported). +# Digital transfers and monitoring can be done as well if you have +# the digital daughterboard (see the section on using the S/PDIF port +# for more information). +# +# Support for the Turtle Beach MultiSound Hurricane architecture is +# composed of the following modules (these can also operate compiled +# into the kernel): +# +# snd-msnd-lib - MultiSound base (requires snd) +# +# snd-msnd-classic - Base audio/mixer support for Classic, Monetery and +# Tahiti cards +# +# snd-msnd-pinnacle - Base audio/mixer support for Pinnacle and Fiji cards +# +# +# Important Notes - Read Before Using +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# The firmware files are not included (may change in future). You +# must obtain these images from Turtle Beach (they are included in +# the MultiSound Development Kits), and place them in /etc/sound for +# example, and give the full paths in the Linux configuration. If +# you are compiling in support for the MultiSound driver rather than +# using it as a module, these firmware files must be accessible +# during kernel compilation. +# +# Please note these files must be binary files, not assembler. See +# the section later in this document for instructions to obtain these +# files. +# +# +# Configuring Card Resources +# ~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# ** This section is very important, as your card may not work at all +# or your machine may crash if you do not do this correctly. ** +# +# * Classic/Monterey/Tahiti +# +# These cards are configured through the driver snd-msnd-classic. You must +# know the io port, then the driver will select the irq and memory resources +# on the card. It is up to you to know if these are free locations or now, +# a conflict can lock the machine up. +# +# * Pinnacle/Fiji +# +# The Pinnacle and Fiji cards have an extra config port, either +# 0x250, 0x260 or 0x270. This port can be disabled to have the card +# configured strictly through PnP, however you lose the ability to +# access the IDE controller and joystick devices on this card when +# using PnP. The included pinnaclecfg program in this shell archive +# can be used to configure the card in non-PnP mode, and in PnP mode +# you can use isapnptools. These are described briefly here. +# +# pinnaclecfg is not required; you can use the snd-msnd-pinnacle module +# to fully configure the card as well. However, pinnaclecfg can be +# used to change the resource values of a particular device after the +# snd-msnd-pinnacle module has been loaded. If you are compiling the +# driver into the kernel, you must set these values during compile +# time, however other peripheral resource values can be changed with +# the pinnaclecfg program after the kernel is loaded. +# +# +# *** PnP mode +# +# Use pnpdump to obtain a sample configuration if you can; I was able +# to obtain one with the command `pnpdump 1 0x203' -- this may vary +# for you (running pnpdump by itself did not work for me). Then, +# edit this file and use isapnp to uncomment and set the card values. +# Use these values when inserting the snd-msnd-pinnacle module. Using +# this method, you can set the resources for the DSP and the Kurzweil +# synth (Pinnacle). Since Linux does not directly support PnP +# devices, you may have difficulty when using the card in PnP mode +# when it the driver is compiled into the kernel. Using non-PnP mode +# is preferable in this case. +# +# Here is an example mypinnacle.conf for isapnp that sets the card to +# io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil +# synth to 0x330 and irq 9 (may need editing for your system): +# +# (READPORT 0x0203) +# (CSN 2) +# (IDENTIFY *) +# +# # DSP +# (CONFIGURE BVJ0440/-1 (LD 0 +# (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000)) +# (ACT Y))) +# +# # Kurzweil Synth (Pinnacle Only) +# (CONFIGURE BVJ0440/-1 (LD 1 +# (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E))) +# (ACT Y))) +# +# (WAITFORKEY) +# +# +# *** Non-PnP mode +# +# The second way is by running the card in non-PnP mode. This +# actually has some advantages in that you can access some other +# devices on the card, such as the joystick and IDE controller. To +# configure the card, unpack this shell archive and build the +# pinnaclecfg program. Using this program, you can assign the +# resource values to the card's devices, or disable the devices. As +# an alternative to using pinnaclecfg, you can specify many of the +# configuration values when loading the snd-msnd-pinnacle module (or +# during kernel configuration when compiling the driver into the +# kernel). +# +# If you specify cfg=0x250 for the snd-msnd-pinnacle module, it +# automatically configure the card to the given io, irq and memory +# values using that config port (the config port is jumper selectable +# on the card to 0x250, 0x260 or 0x270). +# +# See the `snd-msnd-pinnacle Additional Options' section below for more +# information on these parameters (also, if you compile the driver +# directly into the kernel, these extra parameters can be useful +# here). +# +# +# ** It is very easy to cause problems in your machine if you choose a +# resource value which is incorrect. ** +# +# +# Examples +# ~~~~~~~~ +# +# * MultiSound Classic/Monterey/Tahiti: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-classic io=0x290 irq=7 mem=0xd0000 +# +# * MultiSound Pinnacle in PnP mode: +# +# modprobe snd +# insmod snd-msnd-lib +# isapnp mypinnacle.conf +# insmod snd-msnd-pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values +# +# * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port, +# one of 0x250, 0x260 or 0x270): +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP +# mode, add the following (assumes you did `isapnp mypinnacle.conf'): +# +# insmod snd +# insmod mpu401 io=0x330 irq=9 <-- match mypinnacle.conf values +# +# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP +# mode, add the following. Note how we first configure the peripheral's +# resources, _then_ install a Linux driver for it: +# +# insmod snd +# pinnaclecfg 0x250 mpu 0x330 9 +# insmod mpu401 io=0x330 irq=9 +# +# -- OR you can use the following sequence without pinnaclecfg in non-PnP mode: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9 +# insmod snd +# insmod mpu401 io=0x330 irq=9 +# +# * To setup the joystick port on the Pinnacle in non-PnP mode (though +# you have to find the actual Linux joystick driver elsewhere), you +# can use pinnaclecfg: +# +# pinnaclecfg 0x250 joystick 0x200 +# +# -- OR you can configure this using snd-msnd-pinnacle with the following: +# +# modprobe snd +# insmod snd-msnd-lib +# insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200 +# +# +# snd-msnd-classic, snd-msnd-pinnacle Required Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If the following options are not given, the module will not load. +# Examine the kernel message log for informative error messages. +# WARNING--probing isn't supported so try to make sure you have the +# correct shared memory area, otherwise you may experience problems. +# +# io I/O base of DSP, e.g. io=0x210 +# irq IRQ number, e.g. irq=5 +# mem Shared memory area, e.g. mem=0xd8000 +# +# +# snd-msnd-classic, snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# fifosize The digital audio FIFOs, in kilobytes. If not +# specified, the default will be used. Increasing +# this value will reduce the chance of a FIFO +# underflow at the expense of increasing overall +# latency. For example, fifosize=512 will +# allocate 512kB read and write FIFOs (1MB total). +# While this may reduce dropouts, a heavy machine +# load will undoubtedly starve the FIFO of data +# and you will eventually get dropouts. One +# option is to alter the scheduling priority of +# the playback process, using `nice' or some form +# of POSIX soft real-time scheduling. +# +# calibrate_signal Setting this to one calibrates the ADCs to the +# signal, zero calibrates to the card (defaults +# to zero). +# +# +# snd-msnd-pinnacle Additional Options +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# digital Specify digital=1 to enable the S/PDIF input +# if you have the digital daughterboard +# adapter. This will enable access to the +# DIGITAL1 input for the soundcard in the mixer. +# Some mixer programs might have trouble setting +# the DIGITAL1 source as an input. If you have +# trouble, you can try the setdigital.c program +# at the bottom of this document. +# +# cfg Non-PnP configuration port for the Pinnacle +# and Fiji (typically 0x250, 0x260 or 0x270, +# depending on the jumper configuration). If +# this option is omitted, then it is assumed +# that the card is in PnP mode, and that the +# specified DSP resource values are already +# configured with PnP (i.e. it won't attempt to +# do any sort of configuration). +# +# When the Pinnacle is in non-PnP mode, you can use the following +# options to configure particular devices. If a full specification +# for a device is not given, then the device is not configured. Note +# that you still must use a Linux driver for any of these devices +# once their resources are setup (such as the Linux joystick driver, +# or the MPU401 driver from OSS for the Kurzweil synth). +# +# mpu_io I/O port of MPU (on-board Kurzweil synth) +# mpu_irq IRQ of MPU (on-board Kurzweil synth) +# ide_io0 First I/O port of IDE controller +# ide_io1 Second I/O port of IDE controller +# ide_irq IRQ IDE controller +# joystick_io I/O port of joystick +# +# +# Obtaining and Creating Firmware Files +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# For the Classic/Tahiti/Monterey +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach: +# +# ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip +# +# When unzipped, unzip the file named MsndFiles.zip. Then copy the +# following firmware files to /etc/sound (note the file renaming): +# +# cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin +# cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin +# +# When configuring the Linux kernel, specify /etc/sound/msndinit.bin and +# /etc/sound/msndperm.bin for the two firmware files (Linux kernel +# versions older than 2.2 do not ask for firmware paths, and are +# hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# For the Pinnacle/Fiji +# ~~~~~~~~~~~~~~~~~~~~~ +# +# Download to /tmp and unzip the following file from Turtle Beach (be +# sure to use the entire URL; some have had trouble navigating to the +# URL): +# +# ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip +# +# Unpack this shell archive, and run make in the created directory +# (you need a C compiler and flex to build the utilities). This +# should give you the executables conv, pinnaclecfg and setdigital. +# conv is only used temporarily here to create the firmware files, +# while pinnaclecfg is used to configure the Pinnacle or Fiji card in +# non-PnP mode, and setdigital can be used to set the S/PDIF input on +# the mixer (pinnaclecfg and setdigital should be copied to a +# convenient place, possibly run during system initialization). +# +# To generating the firmware files with the `conv' program, we create +# the binary firmware files by doing the following conversion +# (assuming the archive unpacked into a directory named PINNDDK): +# +# ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin +# ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin +# +# The conv (and conv.l) program is not needed after conversion and can +# be safely deleted. Then, when configuring the Linux kernel, specify +# /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two +# firmware files (Linux kernel versions older than 2.2 do not ask for +# firmware paths, and are hardcoded to /etc/sound). +# +# If you are compiling the driver into the kernel, these files must +# be accessible during compilation, but will not be needed later. +# The files must remain, however, if the driver is used as a module. +# +# +# Using Digital I/O with the S/PDIF Port +# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +# +# If you have a Pinnacle or Fiji with the digital daughterboard and +# want to set it as the input source, you can use this program if you +# have trouble trying to do it with a mixer program (be sure to +# insert the module with the digital=1 option, or say Y to the option +# during compiled-in kernel operation). Upon selection of the S/PDIF +# port, you should be able monitor and record from it. +# +# There is something to note about using the S/PDIF port. Digital +# timing is taken from the digital signal, so if a signal is not +# connected to the port and it is selected as recording input, you +# will find PCM playback to be distorted in playback rate. Also, +# attempting to record at a sampling rate other than the DAT rate may +# be problematic (i.e. trying to record at 8000Hz when the DAT signal +# is 44100Hz). If you have a problem with this, set the recording +# input to analog if you need to record at a rate other than that of +# the DAT rate. +# +# +# -- Shell archive attached below, just run `sh MultiSound' to extract. +# Contains Pinnacle/Fiji utilities to convert firmware, configure +# in non-PnP mode, and select the DIGITAL1 input for the mixer. +# +# +#!/bin/sh +# This is a shell archive (produced by GNU sharutils 4.2). +# To extract the files from this archive, save it to some FILE, remove +# everything before the `!/bin/sh' line above, then type `sh FILE'. +# +# Made on 1998-12-04 10:07 EST by <andrewtv@ztransform.velsoft.com>. +# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'. +# +# Existing files will *not* be overwritten unless `-c' is specified. +# +# This shar contains: +# length mode name +# ------ ---------- ------------------------------------------ +# 2064 -rw-rw-r-- MultiSound.d/setdigital.c +# 10224 -rw-rw-r-- MultiSound.d/pinnaclecfg.c +# 106 -rw-rw-r-- MultiSound.d/Makefile +# 146 -rw-rw-r-- MultiSound.d/conv.l +# 1491 -rw-rw-r-- MultiSound.d/msndreset.c +# +save_IFS="${IFS}" +IFS="${IFS}:" +gettext_dir=FAILED +locale_dir=FAILED +first_param="$1" +for dir in $PATH +do + if test "$gettext_dir" = FAILED && test -f $dir/gettext \ + && ($dir/gettext --version >/dev/null 2>&1) + then + set `$dir/gettext --version 2>&1` + if test "$3" = GNU + then + gettext_dir=$dir + fi + fi + if test "$locale_dir" = FAILED && test -f $dir/shar \ + && ($dir/shar --print-text-domain-dir >/dev/null 2>&1) + then + locale_dir=`$dir/shar --print-text-domain-dir` + fi +done +IFS="$save_IFS" +if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED +then + echo=echo +else + TEXTDOMAINDIR=$locale_dir + export TEXTDOMAINDIR + TEXTDOMAIN=sharutils + export TEXTDOMAIN + echo="$gettext_dir/gettext -s" +fi +touch -am 1231235999 $$.touch >/dev/null 2>&1 +if test ! -f 1231235999 && test -f $$.touch; then + shar_touch=touch +else + shar_touch=: + echo + $echo 'WARNING: not restoring timestamps. Consider getting and' + $echo "installing GNU \`touch', distributed in GNU File Utilities..." + echo +fi +rm -f 1231235999 $$.touch +# +if mkdir _sh01426; then + $echo 'x -' 'creating lock directory' +else + $echo 'failed to create lock directory' + exit 1 +fi +# ============= MultiSound.d/setdigital.c ============== +if test ! -d 'MultiSound.d'; then + $echo 'x -' 'creating directory' 'MultiSound.d' + mkdir 'MultiSound.d' +fi +if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' && +/********************************************************************* +X * +X * setdigital.c - sets the DIGITAL1 input for a mixer +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +X +int main(int argc, char *argv[]) +{ +X int fd; +X unsigned long recmask, recsrc; +X +X if (argc != 2) { +X fprintf(stderr, "usage: setdigital <mixer device>\n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) { +X fprintf(stderr, "error: ioctl read recording mask failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X if (!(recmask & SOUND_MASK_DIGITAL1)) { +X fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n"); +X close(fd); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl read recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X recsrc |= SOUND_MASK_DIGITAL1; +X +X if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) { +X fprintf(stderr, "error: ioctl write recording source failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' && + chmod 0664 'MultiSound.d/setdigital.c' || + $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed' +e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`" + test 2064 -eq "$shar_count" || + $echo 'MultiSound.d/setdigital.c:' 'original size' '2064,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/pinnaclecfg.c ============== +if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' && +/********************************************************************* +X * +X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program +X * +X * This is for NON-PnP mode only. For PnP mode, use isapnptools. +X * +X * This is Linux-specific, and must be run with root permissions. +X * +X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <asm/types.h> +#include <sys/io.h> +X +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 +X +#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((BYTE)(w)) +#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8))) +X +typedef __u8 BYTE; +typedef __u16 USHORT; +typedef __u16 WORD; +X +static int config_port = -1; +X +static int msnd_write_cfg(int cfg, int reg, int value) +{ +X outb(reg, cfg); +X outb(value, cfg + 1); +X if (value != inb(cfg + 1)) { +X fprintf(stderr, "error: msnd_write_cfg: I/O error\n"); +X return -EIO; +X } +X return 0; +} +X +static int msnd_read_cfg(int cfg, int reg) +{ +X outb(reg, cfg); +X return inb(cfg + 1); +} +X +static int msnd_write_cfg_io0(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io0(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO), +X msnd_read_cfg(cfg, IREG_IO0_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_io1(int cfg, int num, WORD io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_io1(int cfg, int num, WORD *io) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO), +X msnd_read_cfg(cfg, IREG_IO1_BASEHI)); +X +X return 0; +} +X +static int msnd_write_cfg_irq(int cfg, int num, WORD irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_irq(int cfg, int num, WORD *irq) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER); +X +X return 0; +} +X +static int msnd_write_cfg_mem(int cfg, int num, int mem) +{ +X WORD wmem; +X +X mem >>= 8; +X mem &= 0xfff; +X wmem = (WORD)mem; +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) +X return -EIO; +X if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_mem(int cfg, int num, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X +X *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO), +X msnd_read_cfg(cfg, IREG_MEMBASEHI)); +X *mem <<= 8; +X +X return 0; +} +X +static int msnd_activate_logical(int cfg, int num) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) +X return -EIO; +X return 0; +} +X +static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_write_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_write_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_write_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_write_cfg_mem(cfg, num, mem)) +X return -EIO; +X if (msnd_activate_logical(cfg, num)) +X return -EIO; +X return 0; +} +X +static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem) +{ +X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) +X return -EIO; +X if (msnd_read_cfg_io0(cfg, num, io0)) +X return -EIO; +X if (msnd_read_cfg_io1(cfg, num, io1)) +X return -EIO; +X if (msnd_read_cfg_irq(cfg, num, irq)) +X return -EIO; +X if (msnd_read_cfg_mem(cfg, num, mem)) +X return -EIO; +X return 0; +} +X +static void usage(void) +{ +X fprintf(stderr, +X "\n" +X "pinnaclecfg 1.0\n" +X "\n" +X "usage: pinnaclecfg <config port> [device config]\n" +X "\n" +X "This is for use with the card in NON-PnP mode only.\n" +X "\n" +X "Available devices (not all available for Fiji):\n" +X "\n" +X " Device Description\n" +X " -------------------------------------------------------------------\n" +X " reset Reset all devices (i.e. disable)\n" +X " show Display current device configurations\n" +X "\n" +X " dsp <io> <irq> <mem> Audio device\n" +X " mpu <io> <irq> Internal Kurzweil synth\n" +X " ide <io0> <io1> <irq> On-board IDE controller\n" +X " joystick <io> Joystick port\n" +X "\n"); +X exit(1); +} +X +static int cfg_reset(void) +{ +X int i; +X +X for (i = 0; i < 4; ++i) +X msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0); +X +X return 0; +} +X +static int cfg_show(void) +{ +X int i; +X int count = 0; +X +X for (i = 0; i < 4; ++i) { +X WORD io0, io1, irq; +X int mem; +X msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem); +X switch (i) { +X case 0: +X if (io0 || irq || mem) { +X printf("dsp 0x%x %d 0x%x\n", io0, irq, mem); +X ++count; +X } +X break; +X case 1: +X if (io0 || irq) { +X printf("mpu 0x%x %d\n", io0, irq); +X ++count; +X } +X break; +X case 2: +X if (io0 || io1 || irq) { +X printf("ide 0x%x 0x%x %d\n", io0, io1, irq); +X ++count; +X } +X break; +X case 3: +X if (io0) { +X printf("joystick 0x%x\n", io0); +X ++count; +X } +X break; +X } +X } +X +X if (count == 0) +X fprintf(stderr, "no devices configured\n"); +X +X return 0; +} +X +static int cfg_dsp(int argc, char *argv[]) +{ +X int io, irq, mem; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1 || +X sscanf(argv[2], "0x%x", &mem) != 1) +X usage(); +X +X if (!(io == 0x290 || +X io == 0x260 || +X io == 0x250 || +X io == 0x240 || +X io == 0x230 || +X io == 0x220 || +X io == 0x210 || +X io == 0x3e0)) { +X fprintf(stderr, "error: io must be one of " +X "210, 220, 230, 240, 250, 260, 290, or 3E0\n"); +X usage(); +X } +X +X if (!(irq == 5 || +X irq == 7 || +X irq == 9 || +X irq == 10 || +X irq == 11 || +X irq == 12)) { +X fprintf(stderr, "error: irq must be one of " +X "5, 7, 9, 10, 11 or 12\n"); +X usage(); +X } +X +X if (!(mem == 0xb0000 || +X mem == 0xc8000 || +X mem == 0xd0000 || +X mem == 0xd8000 || +X mem == 0xe0000 || +X mem == 0xe8000)) { +X fprintf(stderr, "error: mem must be one of " +X "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); +X usage(); +X } +X +X return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem); +} +X +static int cfg_mpu(int argc, char *argv[]) +{ +X int io, irq; +X +X if (argc < 2 || +X sscanf(argv[0], "0x%x", &io) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0); +} +X +static int cfg_ide(int argc, char *argv[]) +{ +X int io0, io1, irq; +X +X if (argc < 3 || +X sscanf(argv[0], "0x%x", &io0) != 1 || +X sscanf(argv[0], "0x%x", &io1) != 1 || +X sscanf(argv[1], "%d", &irq) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0); +} +X +static int cfg_joystick(int argc, char *argv[]) +{ +X int io; +X +X if (argc < 1 || +X sscanf(argv[0], "0x%x", &io) != 1) +X usage(); +X +X return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0); +} +X +int main(int argc, char *argv[]) +{ +X char *device; +X int rv = 0; +X +X --argc; ++argv; +X +X if (argc < 2) +X usage(); +X +X sscanf(argv[0], "0x%x", &config_port); +X if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) { +X fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n"); +X exit(1); +X } +X if (ioperm(config_port, 2, 1)) { +X perror("ioperm"); +X fprintf(stderr, "note: pinnaclecfg must be run as root\n"); +X exit(1); +X } +X device = argv[1]; +X +X argc -= 2; argv += 2; +X +X if (strcmp(device, "reset") == 0) +X rv = cfg_reset(); +X else if (strcmp(device, "show") == 0) +X rv = cfg_show(); +X else if (strcmp(device, "dsp") == 0) +X rv = cfg_dsp(argc, argv); +X else if (strcmp(device, "mpu") == 0) +X rv = cfg_mpu(argc, argv); +X else if (strcmp(device, "ide") == 0) +X rv = cfg_ide(argc, argv); +X else if (strcmp(device, "joystick") == 0) +X rv = cfg_joystick(argc, argv); +X else { +X fprintf(stderr, "error: unknown device %s\n", device); +X usage(); +X } +X +X if (rv) +X fprintf(stderr, "error: device configuration failed\n"); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' && + chmod 0664 'MultiSound.d/pinnaclecfg.c' || + $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed' +366bdf27f0db767a3c7921d0a6db20fe MultiSound.d/pinnaclecfg.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`" + test 10224 -eq "$shar_count" || + $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10224,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/Makefile ============== +if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' && +CC = gcc +CFLAGS = -O +PROGS = setdigital msndreset pinnaclecfg conv +X +all: $(PROGS) +X +clean: +X rm -f $(PROGS) +SHAR_EOF + $shar_touch -am 1204092398 'MultiSound.d/Makefile' && + chmod 0664 'MultiSound.d/Makefile' || + $echo 'restore of' 'MultiSound.d/Makefile' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/Makefile:' 'MD5 check failed' +76ca8bb44e3882edcf79c97df6c81845 MultiSound.d/Makefile +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`" + test 106 -eq "$shar_count" || + $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/conv.l ============== +if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' && +%% +[ \n\t,\r] +\;.* +DB +[0-9A-Fa-f]+H { int n; sscanf(yytext, "%xH", &n); printf("%c", n); } +%% +int yywrap() { return 1; } +void main() { yylex(); } +SHAR_EOF + $shar_touch -am 0828231798 'MultiSound.d/conv.l' && + chmod 0664 'MultiSound.d/conv.l' || + $echo 'restore of' 'MultiSound.d/conv.l' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/conv.l:' 'MD5 check failed' +d2411fc32cd71a00dcdc1f009e858dd2 MultiSound.d/conv.l +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`" + test 146 -eq "$shar_count" || + $echo 'MultiSound.d/conv.l:' 'original size' '146,' 'current size' "$shar_count!" + fi +fi +# ============= MultiSound.d/msndreset.c ============== +if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then + $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)' +else + $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)' + sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' && +/********************************************************************* +X * +X * msndreset.c - resets the MultiSound card +X * +X * Copyright (C) 1998 Andrew Veliath +X * +X * This program is free software; you can redistribute it and/or modify +X * it under the terms of the GNU General Public License as published by +X * the Free Software Foundation; either version 2 of the License, or +X * (at your option) any later version. +X * +X * This program is distributed in the hope that it will be useful, +X * but WITHOUT ANY WARRANTY; without even the implied warranty of +X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +X * GNU General Public License for more details. +X * +X * You should have received a copy of the GNU General Public License +X * along with this program; if not, write to the Free Software +X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +X * +X ********************************************************************/ +X +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> +X +int main(int argc, char *argv[]) +{ +X int fd; +X +X if (argc != 2) { +X fprintf(stderr, "usage: msndreset <mixer device>\n"); +X exit(1); +X } +X +X if ((fd = open(argv[1], O_RDWR)) < 0) { +X perror(argv[1]); +X exit(1); +X } +X +X if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) { +X fprintf(stderr, "error: msnd ioctl reset failed\n"); +X perror("ioctl"); +X close(fd); +X exit(1); +X } +X +X close(fd); +X +X return 0; +} +SHAR_EOF + $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' && + chmod 0664 'MultiSound.d/msndreset.c' || + $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed' + if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ + && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then + md5sum -c << SHAR_EOF >/dev/null 2>&1 \ + || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed' +c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c +SHAR_EOF + else + shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`" + test 1491 -eq "$shar_count" || + $echo 'MultiSound.d/msndreset.c:' 'original size' '1491,' 'current size' "$shar_count!" + fi +fi +rm -fr _sh01426 +exit 0 diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst new file mode 100644 index 000000000..2ce41d382 --- /dev/null +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -0,0 +1,373 @@ +=========================================== +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: + + +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. + + +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). + + +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) + + + +MANUALS/PATENTS +=============== + +ftp://opensource.creative.com/pub/doc +------------------------------------- + +LM4545.pdf + AC97 Codec +m2049.pdf + The EMU10K1 Digital Audio Processor +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + +WO 9901813 (A1) + Audio Effects Processor with multiple asynchronous streams + (Jan. 14, 1999) + +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 (https://www.uspto.gov/) +----------------------------------- + +US 5925841 + Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + +US 5928342 + Audio Effects Processor integrated on a single chip + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + (Jul. 27, 1999) + +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 + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + (Oct. 24, 2000) + +US 6151670 + Method for conserving memory storage using a + pool of short term memory registers + (Nov. 21, 2000) + +US 6195715 + Interrupt control for multiple programs communicating with + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register + (Feb. 27, 2001) diff --git a/Documentation/sound/cards/serial-u16550.rst b/Documentation/sound/cards/serial-u16550.rst new file mode 100644 index 000000000..197aeacea --- /dev/null +++ b/Documentation/sound/cards/serial-u16550.rst @@ -0,0 +1,93 @@ +=================================== +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/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst new file mode 100644 index 000000000..6ee993d45 --- /dev/null +++ b/Documentation/sound/cards/via82xx-mixer.rst @@ -0,0 +1,8 @@ +============= +VIA82xx mixer +============= + +On many VIA82xx boards, the ``Input Source Select`` mixer control does not work. +Setting it to ``Input2`` on such boards will cause recording to hang, or fail +with EIO (input/output error) via OSS emulation. This control should be left +at ``Input1`` for such cards. diff --git a/Documentation/sound/designs/channel-mapping-api.rst b/Documentation/sound/designs/channel-mapping-api.rst new file mode 100644 index 000000000..58e6312a4 --- /dev/null +++ b/Documentation/sound/designs/channel-mapping-api.rst @@ -0,0 +1,164 @@ +============================ +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/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst new file mode 100644 index 000000000..935f325db --- /dev/null +++ b/Documentation/sound/designs/compress-offload.rst @@ -0,0 +1,328 @@ +========================= +ALSA Compress-Offload API +========================= + +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) + +State Machine +============= + +The compressed audio stream state machine is described below :: + + +----------+ + | | + | OPEN | + | | + +----------+ + | + | + | compr_set_params() + | + v + compr_free() +----------+ + +------------------------------------| | + | | SETUP | + | +-------------------------| |<-------------------------+ + | | compr_write() +----------+ | + | | ^ | + | | | compr_drain_notify() | + | | | or | + | | | compr_stop() | + | | | | + | | +----------+ | + | | | | | + | | | DRAIN | | + | | | | | + | | +----------+ | + | | ^ | + | | | | + | | | compr_drain() | + | | | | + | v | | + | +----------+ +----------+ | + | | | compr_start() | | compr_stop() | + | | PREPARE |------------------->| RUNNING |--------------------------+ + | | | | | | + | +----------+ +----------+ | + | | | ^ | + | |compr_free() | | | + | | compr_pause() | | compr_resume() | + | | | | | + | v v | | + | +----------+ +----------+ | + | | | | | compr_stop() | + +--->| FREE | | PAUSE |---------------------------+ + | | | | + +----------+ +----------+ + + +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) + +Gapless Playback SM +=================== + +For Gapless, we move from running state to partial drain and back, along +with setting of meta_data and signalling for next track :: + + + +----------+ + compr_drain_notify() | | + +------------------------>| RUNNING | + | | | + | +----------+ + | | + | | + | | compr_next_track() + | | + | V + | +----------+ + | | | + | |NEXT_TRACK| + | | | + | +----------+ + | | + | | + | | compr_partial_drain() + | | + | V + | +----------+ + | | | + +------------------------ | PARTIAL_ | + | DRAIN | + +----------+ + +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/designs/control-names.rst b/Documentation/sound/designs/control-names.rst new file mode 100644 index 000000000..765ff9b5b --- /dev/null +++ b/Documentation/sound/designs/control-names.rst @@ -0,0 +1,142 @@ +=========================== +Standard ALSA Control Names +=========================== + +This document describes standard names of mixer controls. + +Standard Syntax +--------------- +Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION + + +DIRECTION +~~~~~~~~~ +================ =============== +<nothing> both directions +Playback one direction +Capture one direction +Bypass Playback one direction +Bypass Capture one direction +================ =============== + +FUNCTION +~~~~~~~~ +======== ================================= +Switch on/off switch +Volume amplifier +Route route control, hardware specific +======== ================================= + +CHANNEL +~~~~~~~ +============ ================================================== +<nothing> channel independent, or applies to all channels +Front front left/right channels +Surround rear left/right in 4.0/5.1 surround +CLFE C/LFE channels +Center center channel +LFE LFE channel +Side side left/right for 7.1 surround +============ ================================================== + +LOCATION (Physical location of source) +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +============ ===================== +Front front position +Rear rear position +Dock on docking station +Internal 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/designs/index.rst b/Documentation/sound/designs/index.rst new file mode 100644 index 000000000..1eb08e7ba --- /dev/null +++ b/Documentation/sound/designs/index.rst @@ -0,0 +1,17 @@ +Designs and Implementations +=========================== + +.. toctree:: + :maxdepth: 2 + + control-names + channel-mapping-api + compress-offload + timestamping + jack-controls + tracepoints + procfile + powersave + oss-emulation + seq-oss + jack-injection diff --git a/Documentation/sound/designs/jack-controls.rst b/Documentation/sound/designs/jack-controls.rst new file mode 100644 index 000000000..ae25b1531 --- /dev/null +++ b/Documentation/sound/designs/jack-controls.rst @@ -0,0 +1,48 @@ +================== +ALSA Jack Controls +================== + +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/designs/jack-injection.rst b/Documentation/sound/designs/jack-injection.rst new file mode 100644 index 000000000..f97905215 --- /dev/null +++ b/Documentation/sound/designs/jack-injection.rst @@ -0,0 +1,166 @@ +============================ +ALSA Jack Software Injection +============================ + +Simple Introduction On Jack Injection +===================================== + +Here jack injection means users could inject plugin or plugout events +to the audio jacks through debugfs interface, it is helpful to +validate ALSA userspace changes. For example, we change the audio +profile switching code in the pulseaudio, and we want to verify if the +change works as expected and if the change introduce the regression, +in this case, we could inject plugin or plugout events to an audio +jack or to some audio jacks, we don't need to physically access the +machine and plug/unplug physical devices to the audio jack. + +In this design, an audio jack doesn't equal to a physical audio jack. +Sometimes a physical audio jack contains multi functions, and the +ALSA driver creates multi ``jack_kctl`` for a ``snd_jack``, here the +``snd_jack`` represents a physical audio jack and the ``jack_kctl`` +represents a function, for example a physical jack has two functions: +headphone and mic_in, the ALSA ASoC driver will build 2 ``jack_kctl`` +for this jack. The jack injection is implemented based on the +``jack_kctl`` instead of ``snd_jack``. + +To inject events to audio jacks, we need to enable the jack injection +via ``sw_inject_enable`` first, once it is enabled, this jack will not +change the state by hardware events anymore, we could inject plugin or +plugout events via ``jackin_inject`` and check the jack state via +``status``, after we finish our test, we need to disable the jack +injection via ``sw_inject_enable`` too, once it is disabled, the jack +state will be restored according to the last reported hardware events +and will change by future hardware events. + +The Layout of Jack Injection Interface +====================================== + +If users enable the SND_JACK_INJECTION_DEBUG in the kernel, the audio +jack injection interface will be created as below: +:: + + $debugfs_mount_dir/sound + |-- card0 + |-- |-- HDMI_DP_pcm_10_Jack + |-- |-- |-- jackin_inject + |-- |-- |-- kctl_id + |-- |-- |-- mask_bits + |-- |-- |-- status + |-- |-- |-- sw_inject_enable + |-- |-- |-- type + ... + |-- |-- HDMI_DP_pcm_9_Jack + |-- |-- jackin_inject + |-- |-- kctl_id + |-- |-- mask_bits + |-- |-- status + |-- |-- sw_inject_enable + |-- |-- type + |-- card1 + |-- HDMI_DP_pcm_5_Jack + |-- |-- jackin_inject + |-- |-- kctl_id + |-- |-- mask_bits + |-- |-- status + |-- |-- sw_inject_enable + |-- |-- type + ... + |-- Headphone_Jack + |-- |-- jackin_inject + |-- |-- kctl_id + |-- |-- mask_bits + |-- |-- status + |-- |-- sw_inject_enable + |-- |-- type + |-- Headset_Mic_Jack + |-- jackin_inject + |-- kctl_id + |-- mask_bits + |-- status + |-- sw_inject_enable + |-- type + +The Explanation Of The Nodes +====================================== + +kctl_id + read-only, get jack_kctl->kctl's id + :: + + sound/card1/Headphone_Jack# cat kctl_id + Headphone Jack + +mask_bits + read-only, get jack_kctl's supported events mask_bits + :: + + sound/card1/Headphone_Jack# cat mask_bits + 0x0001 HEADPHONE(0x0001) + +status + read-only, get jack_kctl's current status + +- headphone unplugged: + + :: + + sound/card1/Headphone_Jack# cat status + Unplugged + +- headphone plugged: + + :: + + sound/card1/Headphone_Jack# cat status + Plugged + +type + read-only, get snd_jack's supported events from type (all supported events on the physical audio jack) + :: + + sound/card1/Headphone_Jack# cat type + 0x7803 HEADPHONE(0x0001) MICROPHONE(0x0002) BTN_3(0x0800) BTN_2(0x1000) BTN_1(0x2000) BTN_0(0x4000) + +sw_inject_enable + read-write, enable or disable injection + +- injection disabled: + + :: + + sound/card1/Headphone_Jack# cat sw_inject_enable + Jack: Headphone Jack Inject Enabled: 0 + +- injection enabled: + + :: + + sound/card1/Headphone_Jack# cat sw_inject_enable + Jack: Headphone Jack Inject Enabled: 1 + +- to enable jack injection: + + :: + + sound/card1/Headphone_Jack# echo 1 > sw_inject_enable + +- to disable jack injection: + + :: + + sound/card1/Headphone_Jack# echo 0 > sw_inject_enable + +jackin_inject + write-only, inject plugin or plugout + +- to inject plugin: + + :: + + sound/card1/Headphone_Jack# echo 1 > jackin_inject + +- to inject plugout: + + :: + + sound/card1/Headphone_Jack# echo 0 > jackin_inject diff --git a/Documentation/sound/designs/oss-emulation.rst b/Documentation/sound/designs/oss-emulation.rst new file mode 100644 index 000000000..e8dcb9633 --- /dev/null +++ b/Documentation/sound/designs/oss-emulation.rst @@ -0,0 +1,336 @@ +============================= +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/designs/powersave.rst b/Documentation/sound/designs/powersave.rst new file mode 100644 index 000000000..138157452 --- /dev/null +++ b/Documentation/sound/designs/powersave.rst @@ -0,0 +1,43 @@ +========================== +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/designs/procfile.rst b/Documentation/sound/designs/procfile.rst new file mode 100644 index 000000000..e9f7e0cbd --- /dev/null +++ b/Documentation/sound/designs/procfile.rst @@ -0,0 +1,238 @@ +========================== +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_SND_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/designs/seq-oss.rst b/Documentation/sound/designs/seq-oss.rst new file mode 100644 index 000000000..e82ffe0e7 --- /dev/null +++ b/Documentation/sound/designs/seq-oss.rst @@ -0,0 +1,371 @@ +=============================== +OSS Sequencer Emulation on ALSA +=============================== + +Copyright (c) 1998,1999 by Takashi Iwai + +ver.0.1.8; Nov. 16, 1999 + +Description +=========== + +This directory contains the OSS sequencer emulation driver on ALSA. Note +that this program is still in the development state. + +What this does - it provides the emulation of the OSS sequencer, access +via ``/dev/sequencer`` and ``/dev/music`` devices. +The most of applications using OSS can run if the appropriate ALSA +sequencer is prepared. + +The following features are emulated by this driver: + +* Normal sequencer and MIDI events: + + They are converted to the ALSA sequencer events, and sent to the + corresponding port. + +* Timer events: + + The timer is not selectable by ioctl. The control rate is fixed to + 100 regardless of HZ. That is, even on Alpha system, a tick is always + 1/100 second. The base rate and tempo can be changed in ``/dev/music``. + +* Patch loading: + + It purely depends on the synth drivers whether it's supported since + the patch loading is realized by callback to the synth driver. + +* I/O controls: + + Most of controls are accepted. Some controls + are dependent on the synth driver, as well as even on original OSS. + +Furthermore, you can find the following advanced features: + +* Better queue mechanism: + + The events are queued before processing them. + +* Multiple applications: + + You can run two or more applications simultaneously (even for OSS + sequencer)! + However, each MIDI device is exclusive - that is, if a MIDI device + is opened once by some application, other applications can't use + it. No such a restriction in synth devices. + +* Real-time event processing: + + The events can be processed in real time without using out of bound + ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed + events will be processed in real-time without queued. To switch off the + real-time mode, send RELTIME 0 event. + +* ``/proc`` interface: + + The status of applications and devices can be shown via + ``/proc/asound/seq/oss`` at any time. In the later version, + configuration will be changed via ``/proc`` interface, too. + + +Installation +============ + +Run configure script with both sequencer support (``--with-sequencer=yes``) +and OSS emulation (``--with-oss=yes``) options. A module ``snd-seq-oss.o`` +will be created. If the synth module of your sound card supports for OSS +emulation (so far, only Emu8000 driver), this module will be loaded +automatically. +Otherwise, you need to load this module manually. + +At beginning, this module probes all the MIDI ports which have been +already connected to the sequencer. Once after that, the creation and deletion +of ports are watched by announcement mechanism of ALSA sequencer. + +The available synth and MIDI devices can be found in proc interface. +Run ``cat /proc/asound/seq/oss``, and check the devices. For example, +if you use an AWE64 card, you'll see like the following: +:: + + OSS sequencer emulation version 0.1.8 + ALSA client number 63 + ALSA receiver port 0 + + Number of applications: 0 + + Number of synth devices: 1 + synth 0: [EMU8000] + type 0x1 : subtype 0x20 : voices 32 + capabilties : ioctl enabled / load_patch enabled + + Number of MIDI devices: 3 + midi 0: [Emu8000 Port-0] ALSA port 65:0 + capability write / opened none + + midi 1: [Emu8000 Port-1] ALSA port 65:1 + capability write / opened none + + midi 2: [0: MPU-401 (UART)] ALSA port 64:0 + capability read/write / opened none + +Note that the device number may be different from the information of +``/proc/asound/oss-devices`` or ones of the original OSS driver. +Use the device number listed in ``/proc/asound/seq/oss`` +to play via OSS sequencer emulation. + +Using Synthesizer Devices +========================= + +Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 +and xmp-1.1.5. You can load samples via ``/dev/sequencer`` like sfxload, +too. + +If the lowlevel driver supports multiple access to synth devices (like +Emu8000 driver), two or more applications are allowed to run at the same +time. + +Using MIDI Devices +================== + +So far, only MIDI output was tested. MIDI input was not checked at all, +but hopefully it will work. Use the device number listed in +``/proc/asound/seq/oss``. +Be aware that these numbers are mostly different from the list in +``/proc/asound/oss-devices``. + +Module Options +============== + +The following module options are available: + +maxqlen + specifies the maximum read/write queue length. This queue is private + for OSS sequencer, so that it is independent from the queue length of ALSA + sequencer. Default value is 1024. + +seq_oss_debug + specifies the debug level and accepts zero (= no debug message) or + positive integer. Default value is 0. + +Queue Mechanism +=============== + +OSS sequencer emulation uses an ALSA priority queue. The +events from ``/dev/sequencer`` are processed and put onto the queue +specified by module option. + +All the events from ``/dev/sequencer`` are parsed at beginning. +The timing events are also parsed at this moment, so that the events may +be processed in real-time. Sending an event ABSTIME 0 switches the operation +mode to real-time mode, and sending an event RELTIME 0 switches it off. +In the real-time mode, all events are dispatched immediately. + +The queued events are dispatched to the corresponding ALSA sequencer +ports after scheduled time by ALSA sequencer dispatcher. + +If the write-queue is full, the application sleeps until a certain amount +(as default one half) becomes empty in blocking mode. The synchronization +to write timing was implemented, too. + +The input from MIDI devices or echo-back events are stored on read FIFO +queue. If application reads ``/dev/sequencer`` in blocking mode, the +process will be awaked. + +Interface to Synthesizer Device +=============================== + +Registration +------------ + +To register an OSS synthesizer device, use snd_seq_oss_synth_register() +function: +:: + + int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, + snd_seq_oss_callback_t *oper, void *private_data) + +The arguments ``name``, ``type``, ``subtype`` and ``nvoices`` +are used for making the appropriate synth_info structure for ioctl. The +return value is an index number of this device. This index must be remembered +for unregister. If registration is failed, -errno will be returned. + +To release this device, call snd_seq_oss_synth_unregister() function: +:: + + int snd_seq_oss_synth_unregister(int index) + +where the ``index`` is the index number returned by register function. + +Callbacks +--------- + +OSS synthesizer devices have capability for sample downloading and ioctls +like sample reset. In OSS emulation, these special features are realized +by using callbacks. The registration argument oper is used to specify these +callbacks. The following callback functions must be defined: +:: + + snd_seq_oss_callback_t: + int (*open)(snd_seq_oss_arg_t *p, void *closure); + int (*close)(snd_seq_oss_arg_t *p); + int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); + int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); + int (*reset)(snd_seq_oss_arg_t *p); + +Except for ``open`` and ``close`` callbacks, they are allowed to be NULL. + +Each callback function takes the argument type ``snd_seq_oss_arg_t`` as the +first argument. +:: + + struct snd_seq_oss_arg_t { + int app_index; + int file_mode; + int seq_mode; + snd_seq_addr_t addr; + void *private_data; + int event_passing; + }; + +The first three fields, ``app_index``, ``file_mode`` and ``seq_mode`` +are initialized by OSS sequencer. The ``app_index`` is the application +index which is unique to each application opening OSS sequencer. The +``file_mode`` is bit-flags indicating the file operation mode. See +``seq_oss.h`` for its meaning. The ``seq_mode`` is sequencer operation +mode. In the current version, only ``SND_OSSSEQ_MODE_SYNTH`` is used. + +The next two fields, ``addr`` and ``private_data``, must be +filled by the synth driver at open callback. The ``addr`` contains +the address of ALSA sequencer port which is assigned to this device. If +the driver allocates memory for ``private_data``, it must be released +in close callback by itself. + +The last field, ``event_passing``, indicates how to translate note-on +/ off events. In ``PROCESS_EVENTS`` mode, the note 255 is regarded +as velocity change, and key pressure event is passed to the port. In +``PASS_EVENTS`` mode, all note on/off events are passed to the port +without modified. ``PROCESS_KEYPRESS`` mode checks the note above 128 +and regards it as key pressure event (mainly for Emu8000 driver). + +Open Callback +------------- + +The ``open`` is called at each time this device is opened by an application +using OSS sequencer. This must not be NULL. Typically, the open callback +does the following procedure: + +#. Allocate private data record. +#. Create an ALSA sequencer port. +#. Set the new port address on ``arg->addr``. +#. Set the private data record pointer on ``arg->private_data``. + +Note that the type bit-flags in port_info of this synth port must NOT contain +``TYPE_MIDI_GENERIC`` +bit. Instead, ``TYPE_SPECIFIC`` should be used. Also, ``CAP_SUBSCRIPTION`` +bit should NOT be included, too. This is necessary to tell it from other +normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, +return -errno. + +Ioctl Callback +-------------- + +The ``ioctl`` callback is called when the sequencer receives device-specific +ioctls. The following two ioctls should be processed by this callback: + +IOCTL_SEQ_RESET_SAMPLES + reset all samples on memory -- return 0 + +IOCTL_SYNTH_MEMAVL + return the available memory size + +FM_4OP_ENABLE + can be ignored usually + +The other ioctls are processed inside the sequencer without passing to +the lowlevel driver. + +Load_Patch Callback +------------------- + +The ``load_patch`` callback is used for sample-downloading. This callback +must read the data on user-space and transfer to each device. Return 0 +if succeeded, and -errno if failed. The format argument is the patch key +in patch_info record. The buf is user-space pointer where patch_info record +is stored. The offs can be ignored. The count is total data size of this +sample data. + +Close Callback +-------------- + +The ``close`` 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. + +Reset Callback +-------------- + +The ``reset`` callback is called when sequencer device is reset or +closed by applications. The callback should turn off the sounds on the +relevant port immediately, and initialize the status of the port. If this +callback is undefined, OSS seq sends a ``HEARTBEAT`` event to the +port. + +Events +====== + +Most of the events are processed by sequencer and translated to the adequate +ALSA sequencer events, so that each synth device can receive by input_event +callback of ALSA sequencer port. The following ALSA events should be +implemented by the driver: + +============= =================== +ALSA event Original OSS events +============= =================== +NOTEON SEQ_NOTEON, MIDI_NOTEON +NOTE SEQ_NOTEOFF, MIDI_NOTEOFF +KEYPRESS MIDI_KEY_PRESSURE +CHANPRESS SEQ_AFTERTOUCH, MIDI_CHN_PRESSURE +PGMCHANGE SEQ_PGMCHANGE, MIDI_PGM_CHANGE +PITCHBEND SEQ_CONTROLLER(CTRL_PITCH_BENDER), + MIDI_PITCH_BEND +CONTROLLER MIDI_CTL_CHANGE, + SEQ_BALANCE (with CTL_PAN) +CONTROL14 SEQ_CONTROLLER +REGPARAM SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE) +SYSEX SEQ_SYSEX +============= =================== + +The most of these behavior can be realized by MIDI emulation driver +included in the Emu8000 lowlevel driver. In the future release, this module +will be independent. + +Some OSS events (``SEQ_PRIVATE`` and ``SEQ_VOLUME`` events) are passed as event +type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte +packets without any modification. The lowlevel driver should process these +events appropriately. + +Interface to MIDI Device +======================== + +Since the OSS emulation probes the creation and deletion of ALSA MIDI +sequencer ports automatically by receiving announcement from ALSA +sequencer, the MIDI devices don't need to be registered explicitly +like synth devices. +However, the MIDI port_info registered to ALSA sequencer must include +a group name ``SND_SEQ_GROUP_DEVICE`` and a capability-bit +``CAP_READ`` or ``CAP_WRITE``. Also, subscription capabilities, +``CAP_SUBS_READ`` or ``CAP_SUBS_WRITE``, must be defined, too. If +these conditions are not satisfied, the port is not registered as OSS +sequencer MIDI device. + +The events via MIDI devices are parsed in OSS sequencer and converted +to the corresponding ALSA sequencer events. The input from MIDI sequencer +is also converted to MIDI byte events by OSS sequencer. This works just +a reverse way of seq_midi module. + +Known Problems / TODO's +======================= + +* Patch loading via ALSA instrument layer is not implemented yet. + diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst new file mode 100644 index 000000000..7c7ecf5db --- /dev/null +++ b/Documentation/sound/designs/timestamping.rst @@ -0,0 +1,215 @@ +===================== +ALSA PCM Timestamping +===================== + +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 timestamping 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 diff --git a/Documentation/sound/designs/tracepoints.rst b/Documentation/sound/designs/tracepoints.rst new file mode 100644 index 000000000..b0a7e3010 --- /dev/null +++ b/Documentation/sound/designs/tracepoints.rst @@ -0,0 +1,172 @@ +=================== +Tracepoints in ALSA +=================== + +2017/07/02 +Takasahi Sakamoto + +Tracepoints in ALSA PCM core +============================ + +ALSA PCM core registers ``snd_pcm`` subsystem to kernel tracepoint system. +This subsystem includes two categories of tracepoints; for state of PCM buffer +and for processing of PCM hardware parameters. These tracepoints are available +when corresponding kernel configurations are enabled. When ``CONFIG_SND_DEBUG`` +is enabled, the latter tracepoints are available. When additional +``SND_PCM_XRUN_DEBUG`` is enabled too, the former trace points are enabled. + +Tracepoints for state of PCM buffer +------------------------------------ + +This category includes four tracepoints; ``hwptr``, ``applptr``, ``xrun`` and +``hw_ptr_error``. + +Tracepoints for processing of PCM hardware parameters +----------------------------------------------------- + +This category includes two tracepoints; ``hw_mask_param`` and +``hw_interval_param``. + +In a design of ALSA PCM core, data transmission is abstracted as PCM substream. +Applications manage PCM substream to maintain data transmission for PCM frames. +Before starting the data transmission, applications need to configure PCM +substream. In this procedure, PCM hardware parameters are decided by +interaction between applications and ALSA PCM core. Once decided, runtime of +the PCM substream keeps the parameters. + +The parameters are described in struct snd_pcm_hw_params. This +structure includes several types of parameters. Applications set preferable +value to these parameters, then execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE +or SNDRV_PCM_IOCTL_HW_PARAMS. The former is used just for refining available +set of parameters. The latter is used for an actual decision of the parameters. + +The struct snd_pcm_hw_params structure has below members: + +``flags`` + Configurable. ALSA PCM core and some drivers handle this flag to select + convenient parameters or change their behaviour. +``masks`` + Configurable. This type of parameter is described in + struct snd_mask and represent mask values. As of PCM protocol + v2.0.13, three types are defined. + + - SNDRV_PCM_HW_PARAM_ACCESS + - SNDRV_PCM_HW_PARAM_FORMAT + - SNDRV_PCM_HW_PARAM_SUBFORMAT +``intervals`` + Configurable. This type of parameter is described in + struct snd_interval and represent values with a range. As of + PCM protocol v2.0.13, twelve types are defined. + + - SNDRV_PCM_HW_PARAM_SAMPLE_BITS + - SNDRV_PCM_HW_PARAM_FRAME_BITS + - SNDRV_PCM_HW_PARAM_CHANNELS + - SNDRV_PCM_HW_PARAM_RATE + - SNDRV_PCM_HW_PARAM_PERIOD_TIME + - SNDRV_PCM_HW_PARAM_PERIOD_SIZE + - SNDRV_PCM_HW_PARAM_PERIOD_BYTES + - SNDRV_PCM_HW_PARAM_PERIODS + - SNDRV_PCM_HW_PARAM_BUFFER_TIME + - SNDRV_PCM_HW_PARAM_BUFFER_SIZE + - SNDRV_PCM_HW_PARAM_BUFFER_BYTES + - SNDRV_PCM_HW_PARAM_TICK_TIME +``rmask`` + Configurable. This is evaluated at ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE only. Applications can select which + mask/interval parameter can be changed by ALSA PCM core. For + SNDRV_PCM_IOCTL_HW_PARAMS, this mask is ignored and all of parameters + are going to be changed. +``cmask`` + Read-only. After returning from ioctl(2), buffer in user space for + struct snd_pcm_hw_params includes result of each operation. + This mask represents which mask/interval parameter is actually changed. +``info`` + Read-only. This represents hardware/driver capabilities as bit flags + with SNDRV_PCM_INFO_XXX. Typically, applications execute ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE to retrieve this flag, then decide candidates + of parameters and execute ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS to + configure PCM substream. +``msbits`` + Read-only. This value represents available bit width in MSB side of + a PCM sample. When a parameter of SNDRV_PCM_HW_PARAM_SAMPLE_BITS was + decided as a fixed number, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_num`` + Read-only. This value represents numerator of sampling rate in fraction + notation. Basically, when a parameter of SNDRV_PCM_HW_PARAM_RATE was + decided as a single value, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_den`` + Read-only. This value represents denominator of sampling rate in + fraction notation. Basically, when a parameter of + SNDRV_PCM_HW_PARAM_RATE was decided as a single value, this value is + also calculated according to it. Else, zero. But this behaviour depends + on implementations in driver side. +``fifo_size`` + Read-only. This value represents the size of FIFO in serial sound + interface of hardware. Basically, each driver can assigns a proper + value to this parameter but some drivers intentionally set zero with + a care of hardware design or data transmission protocol. + +ALSA PCM core handles buffer of struct snd_pcm_hw_params when +applications execute ioctl(2) with SNDRV_PCM_HW_REFINE or SNDRV_PCM_HW_PARAMS. +Parameters in the buffer are changed according to +struct snd_pcm_hardware and rules of constraints in the runtime. The +structure describes capabilities of handled hardware. The rules describes +dependencies on which a parameter is decided according to several parameters. +A rule has a callback function, and drivers can register arbitrary functions +to compute the target parameter. ALSA PCM core registers some rules to the +runtime as a default. + +Each driver can join in the interaction as long as it prepared for two stuffs +in a callback of struct snd_pcm_ops.open. + +1. In the callback, drivers are expected to change a member of + struct snd_pcm_hardware type in the runtime, according to + capacities of corresponding hardware. +2. In the same callback, drivers are also expected to register additional rules + of constraints into the runtime when several parameters have dependencies + due to hardware design. + +The driver can refers to result of the interaction in a callback of +struct snd_pcm_ops.hw_params, however it should not change the +content. + +Tracepoints in this category are designed to trace changes of the +mask/interval parameters. When ALSA PCM core changes them, ``hw_mask_param`` or +``hw_interval_param`` event is probed according to type of the changed parameter. + +ALSA PCM core also has a pretty print format for each of the tracepoints. Below +is an example for ``hw_mask_param``. + +:: + + hw_mask_param: pcmC0D0p 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 + + +Below is an example for ``hw_interval_param``. + +:: + + hw_interval_param: pcmC0D0p 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295] + +The first three fields are common. They represent name of ALSA PCM character +device, rules of constraint and name of the changed parameter, in order. The +field for rules of constraint consists of two sub-fields; index of applied rule +and total number of rules added to the runtime. As an exception, the index 000 +means that the parameter is changed by ALSA PCM core, regardless of the rules. + +The rest of field represent state of the parameter before/after changing. These +fields are different according to type of the parameter. For parameters of mask +type, the fields represent hexadecimal dump of content of the parameter. For +parameters of interval type, the fields represent values of each member of +``empty``, ``integer``, ``openmin``, ``min``, ``max``, ``openmax`` in +struct snd_interval in this order. + +Tracepoints in drivers +====================== + +Some drivers have tracepoints for developers' convenience. For them, please +refer to each documentation or implementation. diff --git a/Documentation/sound/hd-audio/controls.rst b/Documentation/sound/hd-audio/controls.rst new file mode 100644 index 000000000..dbe6483f4 --- /dev/null +++ b/Documentation/sound/hd-audio/controls.rst @@ -0,0 +1,121 @@ +====================================== +HD-Audio Codec-Specific Mixer Controls +====================================== + + +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 Realtek 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/hd-audio/dp-mst.rst b/Documentation/sound/hd-audio/dp-mst.rst new file mode 100644 index 000000000..1617459e3 --- /dev/null +++ b/Documentation/sound/hd-audio/dp-mst.rst @@ -0,0 +1,101 @@ +======================= +HD-Audio DP-MST Support +======================= + +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 + +Pin Initialization +================== +Each pin may have several device entries (virtual pins). On Intel platform, +the device entries number is dynamically changed. If DP MST hub is connected, +it is in DP MST mode, and the device entries number is 3. Otherwise, the +device entries number is 1. + +To simplify the implementation, all the device entries will be initialized +when bootup no matter whether it is in DP MST mode or not. + +Connection list +=============== +DP MST reuses connection list code. The code can be reused because +device entries on the same pin have the same connection list. + +This means DP MST gets the device entry connection list without the +device entry setting. + +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/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst new file mode 100644 index 000000000..6e12de9fc --- /dev/null +++ b/Documentation/sound/hd-audio/index.rst @@ -0,0 +1,11 @@ +HD-Audio +======== + +.. toctree:: + :maxdepth: 2 + + notes + models + controls + dp-mst + realtek-pc-beep diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst new file mode 100644 index 000000000..120430450 --- /dev/null +++ b/Documentation/sound/hd-audio/models.rst @@ -0,0 +1,809 @@ +============================== +HD-Audio Codec-Specific Models +============================== + +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 +6stack-automute + 6-jack with headphone jack detection + +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 +fsc-h270 + Fixups for Fujitsu-Siemens Celsius H270 +fsc-s7110 + Fixups for Fujitsu-Siemens Lifebook S7110 +hp-z200 + Fixups for HP Z200 +tyan + Fixups for Tyan Thunder n6650W +lenovo-3000 + Fixups for Lenovo 3000 +benq + Fixups for Benq ED8 +benq-t31 + Fixups for Benq T31 +bayleybay + Fixups for Intel BayleyBay + +ALC267/268 +========== +inv-dmic + Inverted internal mic workaround +hp-eapd + Disable HP EAPD on NID 0x15 +spdif + Enable SPDIF output on NID 0x1e + +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 +hp-dock-gpio-mic1-led + HP dock with mic LED support +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 +dell-headset3 + Headset jack (without mic-in), and also dock I/O, variant 3 +dell-headset4 + Headset jack (without mic-in), and also dock I/O, variant 4 +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 +tpt440 + Lenovo Thinkpad T440s setup +tpt460 + Lenovo Thinkpad T460/560 setup +tpt470-dock + Lenovo Thinkpad T470 dock setup +dual-codecs + Lenovo laptops with dual codecs +alc700-ref + Intel reference board with ALC700 codec +vaio + Pin fixups for Sony VAIO laptops +dell-m101z + COEF setup for Dell M101z +asus-g73jw + Subwoofer pin fixup for ASUS G73JW +lenovo-eapd + Inversed EAPD setup for Lenovo laptops +sony-hweq + H/W EQ COEF setup for Sony laptops +pcm44k + Fixed PCM 44kHz constraints (for buggy devices) +lifebook + Dock pin fixups for Fujitsu Lifebook +lifebook-extmic + Headset mic fixup for Fujitsu Lifebook +lifebook-hp-pin + Headphone pin fixup for Fujitsu Lifebook +lifebook-u7x7 + Lifebook U7x7 fixups +alc269vb-amic + ALC269VB analog mic pin fixups +alc269vb-dmic + ALC269VB digital mic pin fixups +hp-mute-led-mic1 + Mute LED via Mic1 pin on HP +hp-mute-led-mic2 + Mute LED via Mic2 pin on HP +hp-mute-led-mic3 + Mute LED via Mic3 pin on HP +hp-gpio-mic1 + GPIO + Mic1 pin LED on HP +hp-line1-mic1 + Mute LED via Line1 + Mic1 pins on HP +noshutup + Skip shutup callback +sony-nomic + Headset mic fixup for Sony laptops +aspire-headset-mic + Headset pin fixup for Acer Aspire +asus-x101 + ASUS X101 fixups +acer-ao7xx + Acer AO7xx fixups +acer-aspire-e1 + Acer Aspire E1 fixups +acer-ac700 + Acer AC700 fixups +limit-mic-boost + Limit internal mic boost on Lenovo machines +asus-zenbook + ASUS Zenbook fixups +asus-zenbook-ux31a + ASUS Zenbook UX31A fixups +ordissimo + Ordissimo EVE2 (or Malata PC-B1303) fixups +asus-tx300 + ASUS TX300 fixups +alc283-int-mic + ALC283 COEF setup for Lenovo machines +mono-speakers + Subwoofer and headset fixupes for Dell Inspiron +alc290-subwoofer + Subwoofer fixups for Dell Vostro +thinkpad + Binding with thinkpad_acpi driver for Lenovo machines +dmic-thinkpad + thinkpad_acpi binding + digital mic support +alc255-acer + ALC255 fixups on Acer machines +alc255-asus + ALC255 fixups on ASUS machines +alc255-dell1 + ALC255 fixups on Dell machines +alc255-dell2 + ALC255 fixups on Dell machines, variant 2 +alc293-dell1 + ALC293 fixups on Dell machines +alc283-headset + Headset pin fixups on ALC283 +aspire-v5 + Acer Aspire V5 fixups +hp-gpio4 + GPIO and Mic1 pin mute LED fixups for HP +hp-gpio-led + GPIO mute LEDs on HP +hp-gpio2-hotkey + GPIO mute LED with hot key handling on HP +hp-dock-pins + GPIO mute LEDs and dock support on HP +hp-dock-gpio-mic + GPIO, Mic mute LED and dock support on HP +hp-9480m + HP 9480m fixups +alc288-dell1 + ALC288 fixups on Dell machines +alc288-dell-xps13 + ALC288 fixups on Dell XPS13 +dell-e7x + Dell E7x fixups +alc293-dell + ALC293 fixups on Dell machines +alc298-dell1 + ALC298 fixups on Dell machines +alc298-dell-aio + ALC298 fixups on Dell AIO machines +alc275-dell-xps + ALC275 fixups on Dell XPS models +lenovo-spk-noise + Workaround for speaker noise on Lenovo machines +lenovo-hotkey + Hot-key support via Mic2 pin on Lenovo machines +dell-spk-noise + Workaround for speaker noise on Dell machines +alc255-dell1 + ALC255 fixups on Dell machines +alc295-disable-dac3 + Disable DAC3 routing on ALC295 +alc280-hp-headset + HP Elitebook fixups +alc221-hp-mic + Front mic pin fixup on HP machines +alc298-spk-volume + Speaker pin routing workaround on ALC298 +dell-inspiron-7559 + Dell Inspiron 7559 fixups +ativ-book + Samsung Ativ book 8 fixups +alc221-hp-mic + ALC221 headset fixups on HP machines +alc256-asus-mic + ALC256 fixups on ASUS machines +alc256-asus-aio + ALC256 fixups on ASUS AIO machines +alc233-eapd + ALC233 fixups on ASUS machines +alc294-lenovo-mic + ALC294 Mic pin fixup for Lenovo AIO machines +alc225-wyse + Dell Wyse fixups +alc274-dell-aio + ALC274 fixups on Dell AIO machines +alc255-dummy-lineout + Dell Precision 3930 fixups +alc255-dell-headset + Dell Precision 3630 fixups +alc295-hp-x360 + HP Spectre X360 fixups +alc-sense-combo + Headset button support for Chrome platform +huawei-mbx-stereo + Enable initialization verbs for Huawei MBX stereo speakers; + might be risky, try this at your own risk +alc298-samsung-headphone + Samsung laptops with ALC298 +alc256-samsung-headphone + Samsung laptops with ALC256 + +ALC66x/67x/892 +============== +aspire + Subwoofer pin fixup for Aspire laptops +ideapad + Subwoofer pin fixup for Ideapad laptops +mario + Chromebook mario model fixup +hp-rp5800 + Headphone pin fixup for HP RP5800 +asus-mode1 + ASUS +asus-mode2 + ASUS +asus-mode3 + ASUS +asus-mode4 + ASUS +asus-mode5 + ASUS +asus-mode6 + ASUS +asus-mode7 + ASUS +asus-mode8 + ASUS +zotac-z68 + Front HP fixup for Zotac Z68 +inv-dmic + Inverted internal mic workaround +alc662-headset-multi + Dell headset jack, which can also be used as mic-in (ALC662) +dell-headset-multi + Headset jack, which can also be used as mic-in +alc662-headset + Headset mode support on ALC662 +alc668-headset + Headset mode support on ALC668 +bass16 + Bass speaker fixup on pin 0x16 +bass1a + Bass speaker fixup on pin 0x1a +automute + Auto-mute fixups for ALC668 +dell-xps13 + Dell XPS13 fixups +asus-nx50 + ASUS Nx50 fixups +asus-nx51 + ASUS Nx51 fixups +asus-g751 + ASUS G751 fixups +alc891-headset + Headset mode support on ALC891 +alc891-headset-multi + Dell headset jack, which can also be used as mic-in (ALC891) +acer-veriton + Acer Veriton speaker pin fixup +asrock-mobo + Fix invalid 0x15 / 0x16 pins +usi-headset + Headset support on USI machines +dual-codecs + Lenovo laptops with dual codecs +alc285-hp-amp-init + HP laptops which require speaker amplifier initialization (ALC285) + +ALC680 +====== +N/A + +ALC88x/898/1150/1220 +==================== +abit-aw9d + Pin fixups for Abit AW9D-MAX +lenovo-y530 + Pin fixups for Lenovo Y530 +acer-aspire-7736 + Fixup for Acer Aspire 7736 +asus-w90v + Pin fixup for ASUS W90V +cd + Enable audio CD pin NID 0x1c +no-front-hp + Disable front HP pin NID 0x1b +vaio-tt + Pin fixup for VAIO TT +eee1601 + COEF setups for ASUS Eee 1601 +alc882-eapd + Change EAPD COEF mode on ALC882 +alc883-eapd + Change EAPD COEF mode on ALC883 +gpio1 + Enable GPIO1 +gpio2 + Enable GPIO2 +gpio3 + Enable GPIO3 +alc889-coef + Setup ALC889 COEF +asus-w2jc + Fixups for ASUS W2JC +acer-aspire-4930g + Acer Aspire 4930G/5930G/6530G/6930G/7730G +acer-aspire-8930g + Acer Aspire 8330G/6935G +acer-aspire + Acer Aspire others +macpro-gpio + GPIO setup for Mac Pro +dac-route + Workaround for DAC routing on Acer Aspire +mbp-vref + Vref setup for Macbook Pro +imac91-vref + Vref setup for iMac 9,1 +mba11-vref + Vref setup for MacBook Air 1,1 +mba21-vref + Vref setup for MacBook Air 2,1 +mp11-vref + Vref setup for Mac Pro 1,1 +mp41-vref + Vref setup for Mac Pro 4,1 +inv-dmic + Inverted internal mic workaround +no-primary-hp + VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) +asus-bass + Bass speaker setup for ASUS ET2700 +dual-codecs + ALC1220 dual codecs for Gaming mobos +clevo-p950 + Fixups for Clevo P950 + +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 +============= +cap-mix-amp + Fix max input level on mixer widget +toshiba-p105 + Toshiba P105 quirk +hp-530 + HP 530 quirk + +Conexant 5047 +============= +cap-mix-amp + Fix max input level on mixer widget + +Conexant 5051 +============= +lenovo-x200 + Lenovo X200 quirk + +Conexant 5066 +============= +stereo-dmic + Workaround for inverted stereo digital mic +gpio1 + Enable GPIO1 pin +headphone-mic-pin + Enable headphone mic NID 0x18 without detection +tp410 + Thinkpad T400 & co quirks +thinkpad + Thinkpad mute/mic LED quirk +lemote-a1004 + Lemote A1004 quirk +lemote-a1205 + Lemote A1205 quirk +olpc-xo + OLPC XO quirk +mute-led-eapd + Mute LED control via EAPD +hp-dock + HP dock support +mute-led-gpio + Mute LED control via GPIO +hp-mic-fix + Fix for headset mic pin on HP boxes + +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 D*45* 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 +asus-mobo + Pin configs for ASUS mobo with 5.1/SPDIF out +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 +hp-envy-ts-bass + HP Envy TS bass support +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 +======================== +mbp53 + MacBook Pro 5,3 +mbp55 + MacBook Pro 5,5 +imac27 + IMac 27 Inch +imac27_122 + iMac 12,2 +apple + Generic Apple quirk +mbp101 + MacBookPro 10,1 +mbp81 + MacBookPro 8,1 +mba42 + MacBookAir 4,2 +auto + BIOS setup (default) + +Cirrus Logic CS4208 +=================== +mba6 + MacBook Air 6,1 and 6,2 +gpio0 + Enable GPIO 0 amp +mbp11 + MacBookPro 11,2 +macmini + MacMini 7,1 +auto + BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== +auto + BIOS setup (default) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst new file mode 100644 index 000000000..d118b6fe2 --- /dev/null +++ b/Documentation/sound/hd-audio/notes.rst @@ -0,0 +1,898 @@ +============================= +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: + +* https://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. +``position_fix=5`` is specific to Intel platforms, so far, for Skylake +and onward. It applies the delay calculation for the precise position +reporting. +``position_fix=6`` is to correct the position with the fixed FIFO +size, mainly targeted for the recent AMD controllers. +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/hd-audio/models.rst 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. + +A new style for the model option that was introduced since 5.15 kernel +is to pass the PCI or codec SSID in the form of ``model=XXXX:YYYY`` +where XXXX and YYYY are the sub-vendor and sub-device IDs in hex +numbers, respectively. This is a kind of aliasing to another device; +when this form is given, the driver will refer to that SSID as a +reference to the quirk table. It'd be useful especially when the +target quirk isn't listed in the model table. For example, passing +model=103c:8862 will apply the quirk for HP ProBook 445 G8 (which +isn't found in the model table as of writing) as long as the device is +handled equivalently by the same driver. + + +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 +vmaster (bool) + enable/disable the virtual Master control; 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 following: + +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: + +* https://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: + +* https://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/hd-audio/realtek-pc-beep.rst b/Documentation/sound/hd-audio/realtek-pc-beep.rst new file mode 100644 index 000000000..be47c6f76 --- /dev/null +++ b/Documentation/sound/hd-audio/realtek-pc-beep.rst @@ -0,0 +1,129 @@ +=============================== +Realtek PC Beep Hidden Register +=============================== + +This file documents the "PC Beep Hidden Register", which is present in certain +Realtek HDA codecs and controls a muxer and pair of passthrough mixers that can +route audio between pins but aren't themselves exposed as HDA widgets. As far +as I can tell, these hidden routes are designed to allow flexible PC Beep output +for codecs that don't have mixer widgets in their output paths. Why it's easier +to hide a mixer behind an undocumented vendor register than to just expose it +as a widget, I have no idea. + +Register Description +==================== + +The register is accessed via processing coefficient 0x36 on NID 20h. Bits not +identified below have no discernible effect on my machine, a Dell XPS 13 9350:: + + MSB LSB + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + | |h|S|L| | B |R| | Known bits + +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ + |0|0|1|1| 0x7 |0|0x0|1| 0x7 | Reset value + +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + +1Ah input select (B): 2 bits + When zero, expose the PC Beep line (from the internal beep generator, when + enabled with the Set Beep Generation verb on NID 01h, or else from the + external PCBEEP pin) on the 1Ah pin node. When nonzero, expose the headphone + jack (or possibly Line In on some machines) input instead. If PC Beep is + selected, the 1Ah boost control has no effect. + +Amplify 1Ah loopback, left (L): 1 bit + Amplify the left channel of 1Ah before mixing it into outputs as specified + by h and S bits. Does not affect the level of 1Ah exposed to other widgets. + +Amplify 1Ah loopback, right (R): 1 bit + Amplify the right channel of 1Ah before mixing it into outputs as specified + by h and S bits. Does not affect the level of 1Ah exposed to other widgets. + +Loopback 1Ah to 21h [active low] (h): 1 bit + When zero, mix 1Ah (possibly with amplification, depending on L and R bits) + into 21h (headphone jack on my machine). Mixed signal respects the mute + setting on 21h. + +Loopback 1Ah to 14h (S): 1 bit + When one, mix 1Ah (possibly with amplification, depending on L and R bits) + into 14h (internal speaker on my machine). Mixed signal **ignores** the mute + setting on 14h and is present whenever 14h is configured as an output. + +Path diagrams +============= + +1Ah input selection (DIV is the PC Beep divider set on NID 01h):: + + <Beep generator> <PCBEEP pin> <Headphone jack> + | | | + +--DIV--+--!DIV--+ {1Ah boost control} + | | + +--(b == 0)--+--(b != 0)--+ + | + >1Ah (Beep/Headphone Mic/Line In)< + +Loopback of 1Ah to 21h/14h:: + + <1Ah (Beep/Headphone Mic/Line In)> + | + {amplify if L/R} + | + +-----!h-----+-----S-----+ + | | + {21h mute control} | + | | + >21h (Headphone)< >14h (Internal Speaker)< + +Background +========== + +All Realtek HDA codecs have a vendor-defined widget with node ID 20h which +provides access to a bank of registers that control various codec functions. +Registers are read and written via the standard HDA processing coefficient +verbs (Set/Get Coefficient Index, Set/Get Processing Coefficient). The node is +named "Realtek Vendor Registers" in public datasheets' verb listings and, +apart from that, is entirely undocumented. + +This particular register, exposed at coefficient 0x36 and named in commits from +Realtek, is of note: unlike most registers, which seem to control detailed +amplifier parameters not in scope of the HDA specification, it controls audio +routing which could just as easily have been defined using standard HDA mixer +and selector widgets. + +Specifically, it selects between two sources for the input pin widget with Node +ID (NID) 1Ah: the widget's signal can come either from an audio jack (on my +laptop, a Dell XPS 13 9350, it's the headphone jack, but comments in Realtek +commits indicate that it might be a Line In on some machines) or from the PC +Beep line (which is itself multiplexed between the codec's internal beep +generator and external PCBEEP pin, depending on if the beep generator is +enabled via verbs on NID 01h). Additionally, it can mix (with optional +amplification) that signal onto the 21h and/or 14h output pins. + +The register's reset value is 0x3717, corresponding to PC Beep on 1Ah that is +then amplified and mixed into both the headphones and the speakers. Not only +does this violate the HDA specification, which says that "[a vendor defined +beep input pin] connection may be maintained *only* while the Link reset +(**RST#**) is asserted", it means that we cannot ignore the register if we care +about the input that 1Ah would otherwise expose or if the PCBEEP trace is +poorly shielded and picks up chassis noise (both of which are the case on my +machine). + +Unfortunately, there are lots of ways to get this register configuration wrong. +Linux, it seems, has gone through most of them. For one, the register resets +after S3 suspend: judging by existing code, this isn't the case for all vendor +registers, and it's led to some fixes that improve behavior on cold boot but +don't last after suspend. Other fixes have successfully switched the 1Ah input +away from PC Beep but have failed to disable both loopback paths. On my +machine, this means that the headphone input is amplified and looped back to +the headphone output, which uses the exact same pins! As you might expect, this +causes terrible headphone noise, the character of which is controlled by the +1Ah boost control. (If you've seen instructions online to fix XPS 13 headphone +noise by changing "Headphone Mic Boost" in ALSA, now you know why.) + +The information here has been obtained through black-box reverse engineering of +the ALC256 codec's behavior and is not guaranteed to be correct. It likely +also applies for the ALC255, ALC257, ALC235, and ALC236, since those codecs +seem to be close relatives of the ALC256. (They all share one initialization +function.) Additionally, other codecs like the ALC225 and ALC285 also have this +register, judging by existing fixups in ``patch_realtek.c``, but specific +data (e.g. node IDs, bit positions, pin mappings) for those codecs may differ +from what I've described here. diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst new file mode 100644 index 000000000..4d7d42acf --- /dev/null +++ b/Documentation/sound/index.rst @@ -0,0 +1,20 @@ +=================================== +Linux Sound Subsystem Documentation +=================================== + +.. toctree:: + :maxdepth: 2 + + kernel-api/index + designs/index + soc/index + alsa-configuration + hd-audio/index + cards/index + +.. only:: subproject and html + + Indices + ======= + + * :ref:`genindex` diff --git a/Documentation/sound/kernel-api/alsa-driver-api.rst b/Documentation/sound/kernel-api/alsa-driver-api.rst new file mode 100644 index 000000000..d24c64df7 --- /dev/null +++ b/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -0,0 +1,135 @@ +=================== +The ALSA Driver API +=================== + +Management of Cards and Devices +=============================== + +Card Management +--------------- +.. kernel-doc:: sound/core/init.c + +Device Components +----------------- +.. kernel-doc:: sound/core/device.c + +Module requests and Device File Entries +--------------------------------------- +.. kernel-doc:: sound/core/sound.c + +Memory Management Helpers +------------------------- +.. kernel-doc:: sound/core/memory.c +.. kernel-doc:: sound/core/memalloc.c + + +PCM API +======= + +PCM Core +-------- +.. kernel-doc:: sound/core/pcm.c +.. kernel-doc:: sound/core/pcm_lib.c +.. kernel-doc:: sound/core/pcm_native.c +.. kernel-doc:: include/sound/pcm.h + +PCM Format Helpers +------------------ +.. kernel-doc:: sound/core/pcm_misc.c + +PCM Memory Management +--------------------- +.. kernel-doc:: sound/core/pcm_memory.c + +PCM DMA Engine API +------------------ +.. kernel-doc:: sound/core/pcm_dmaengine.c +.. kernel-doc:: include/sound/dmaengine_pcm.h + +Control/Mixer API +================= + +General Control Interface +------------------------- +.. kernel-doc:: sound/core/control.c + +AC97 Codec API +-------------- +.. kernel-doc:: sound/pci/ac97/ac97_codec.c +.. kernel-doc:: sound/pci/ac97/ac97_pcm.c + +Virtual Master Control API +-------------------------- +.. kernel-doc:: sound/core/vmaster.c +.. kernel-doc:: include/sound/control.h + +MIDI API +======== + +Raw MIDI API +------------ +.. kernel-doc:: sound/core/rawmidi.c + +MPU401-UART API +--------------- +.. kernel-doc:: sound/drivers/mpu401/mpu401_uart.c + +Proc Info API +============= + +Proc Info Interface +------------------- +.. kernel-doc:: sound/core/info.c + +Compress Offload +================ + +Compress Offload API +-------------------- +.. kernel-doc:: sound/core/compress_offload.c +.. kernel-doc:: include/uapi/sound/compress_offload.h +.. kernel-doc:: include/uapi/sound/compress_params.h +.. kernel-doc:: include/sound/compress_driver.h + +ASoC +==== + +ASoC Core API +------------- +.. kernel-doc:: include/sound/soc.h +.. kernel-doc:: sound/soc/soc-core.c +.. kernel-doc:: sound/soc/soc-devres.c +.. kernel-doc:: sound/soc/soc-component.c +.. kernel-doc:: sound/soc/soc-pcm.c +.. kernel-doc:: sound/soc/soc-ops.c +.. kernel-doc:: sound/soc/soc-compress.c + +ASoC DAPM API +------------- +.. kernel-doc:: sound/soc/soc-dapm.c + +ASoC DMA Engine API +------------------- +.. kernel-doc:: sound/soc/soc-generic-dmaengine-pcm.c + +Miscellaneous Functions +======================= + +Hardware-Dependent Devices API +------------------------------ +.. kernel-doc:: sound/core/hwdep.c + +Jack Abstraction Layer API +-------------------------- +.. kernel-doc:: include/sound/jack.h +.. kernel-doc:: sound/core/jack.c +.. kernel-doc:: sound/soc/soc-jack.c + +ISA DMA Helpers +--------------- +.. kernel-doc:: sound/core/isadma.c + +Other Helper Macros +------------------- +.. kernel-doc:: include/sound/core.h +.. kernel-doc:: sound/sound_core.c diff --git a/Documentation/sound/kernel-api/index.rst b/Documentation/sound/kernel-api/index.rst new file mode 100644 index 000000000..d0e6df35b --- /dev/null +++ b/Documentation/sound/kernel-api/index.rst @@ -0,0 +1,8 @@ +ALSA Kernel API Documentation +============================= + +.. toctree:: + :maxdepth: 2 + + alsa-driver-api + writing-an-alsa-driver diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst new file mode 100644 index 000000000..07a620c5c --- /dev/null +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -0,0 +1,4339 @@ +====================== +Writing an ALSA Driver +====================== + +:Author: Takashi Iwai <tiwai@suse.de> + +Preface +======= + +This document describes how to write an `ALSA (Advanced Linux Sound +Architecture) <http://www.alsa-project.org/>`__ driver. The document +focuses mainly on PCI soundcards. In the case of other device types, the +API might be different, too. However, at least the ALSA kernel API is +consistent, and therefore it would be still a bit help for writing them. + +This document targets people who already have enough C language skills +and have basic linux kernel programming knowledge. This document doesn't +explain the general topic of linux kernel coding and doesn't cover +low-level driver implementation details. It only describes the standard +way to write a PCI sound driver on ALSA. + +This document is still a draft version. Any feedback and corrections, +please!! + +File Tree Structure +=================== + +General +------- + +The file tree structure of ALSA driver is depicted below. + +:: + + sound + /core + /oss + /seq + /oss + /include + /drivers + /mpu401 + /opl3 + /i2c + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /soc + /oss + + +core directory +-------------- + +This directory contains the middle layer which is the heart of ALSA +drivers. In this directory, the native ALSA modules are stored. The +sub-directories contain different modules and are dependent upon the +kernel config. + +core/oss +~~~~~~~~ + +The codes for PCM and mixer OSS emulation modules are stored in this +directory. The rawmidi OSS emulation is included in the ALSA rawmidi +code since it's quite small. The sequencer code is stored in +``core/seq/oss`` directory (see `below <core/seq/oss_>`__). + +core/seq +~~~~~~~~ + +This directory and its sub-directories are for the ALSA sequencer. This +directory contains the sequencer core and primary sequencer modules such +like snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when +``CONFIG_SND_SEQUENCER`` is set in the kernel config. + +core/seq/oss +~~~~~~~~~~~~ + +This contains the OSS sequencer emulation codes. + +include directory +----------------- + +This is the place for the public header files of ALSA drivers, which are +to be exported to user-space, or included by several files at different +directories. Basically, the private header files should not be placed in +this directory, but you may still find files there, due to historical +reasons :) + +drivers directory +----------------- + +This directory contains code shared among different drivers on different +architectures. They are hence supposed not to be architecture-specific. +For example, the dummy pcm driver and the serial MIDI driver are found +in this directory. In the sub-directories, there is code for components +which are independent from bus and cpu architectures. + +drivers/mpu401 +~~~~~~~~~~~~~~ + +The MPU401 and MPU401-UART modules are stored here. + +drivers/opl3 and opl4 +~~~~~~~~~~~~~~~~~~~~~ + +The OPL3 and OPL4 FM-synth stuff is found here. + +i2c directory +------------- + +This contains the ALSA i2c components. + +Although there is a standard i2c layer on Linux, ALSA has its own i2c +code for some cards, because the soundcard needs only a simple operation +and the standard i2c API is too complicated for such a purpose. + +synth directory +--------------- + +This contains the synth middle-level modules. + +So far, there is only Emu8000/Emu10k1 synth driver under the +``synth/emux`` sub-directory. + +pci directory +------------- + +This directory and its sub-directories hold the top-level card modules +for PCI soundcards and the code specific to the PCI BUS. + +The drivers compiled from a single file are stored directly in the pci +directory, while the drivers with several source files are stored on +their own sub-directory (e.g. emu10k1, ice1712). + +isa directory +------------- + +This directory and its sub-directories hold the top-level card modules +for ISA soundcards. + +arm, ppc, and sparc directories +------------------------------- + +They are used for top-level card modules which are specific to one of +these architectures. + +usb directory +------------- + +This directory contains the USB-audio driver. In the latest version, the +USB MIDI driver is integrated in the usb-audio driver. + +pcmcia directory +---------------- + +The PCMCIA, especially PCCard drivers will go here. CardBus drivers will +be in the pci directory, because their API is identical to that of +standard PCI cards. + +soc directory +------------- + +This directory contains the codes for ASoC (ALSA System on Chip) +layer including ASoC core, codec and machine drivers. + +oss directory +------------- + +Here contains OSS/Lite codes. +All codes have been deprecated except for dmasound on m68k as of +writing this. + + +Basic Flow for PCI Drivers +========================== + +Outline +------- + +The minimum flow for PCI soundcards is as follows: + +- define the PCI ID table (see the section `PCI Entries`_). + +- create ``probe`` callback. + +- create ``remove`` callback. + +- create a struct pci_driver structure + containing the three pointers above. + +- create an ``init`` function just calling the + :c:func:`pci_register_driver()` to register the pci_driver + table defined above. + +- create an ``exit`` function to call the + :c:func:`pci_unregister_driver()` function. + +Full Code Example +----------------- + +The code example is shown below. Some parts are kept unimplemented at +this moment but will be filled in the next sections. The numbers in the +comment lines of the :c:func:`snd_mychip_probe()` function refer +to details explained in the following section. + +:: + + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> + + /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + + /* definition of the chip-specific record */ + struct mychip { + struct snd_card *card; + /* the rest of the implementation will be in section + * "PCI Resource Management" + */ + }; + + /* chip-specific destructor + * (see "PCI Resource Management") + */ + static int snd_mychip_free(struct mychip *chip) + { + .... /* will be implemented later... */ + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static const struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* check PCI availability here + * (see "PCI Resource Management") + */ + .... + + /* allocate a chip-specific data with zero filled */ + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + /* rest of initialization here; will be implemented + * later, see "PCI Resource Management" + */ + .... + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* constructor -- see "Driver Constructor" sub-section */ + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + static int dev; + struct snd_card *card; + struct mychip *chip; + int err; + + /* (1) */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) */ + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + if (err < 0) + return err; + + /* (3) */ + err = snd_mychip_create(card, pci, &chip); + if (err < 0) + goto error; + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->port, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) + goto error; + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + + error: + snd_card_free(card); + return err; + } + + /* destructor -- see the "Destructor" sub-section */ + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + } + + + +Driver Constructor +------------------ + +The real constructor of PCI drivers is the ``probe`` callback. The +``probe`` callback and other component-constructors which are called +from the ``probe`` callback cannot be used with the ``__init`` prefix +because any PCI device could be a hotplug device. + +In the ``probe`` callback, the following scheme is often used. + +1) Check and increment the device index. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int dev; + .... + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + +where ``enable[dev]`` is the module option. + +Each time the ``probe`` callback is called, check the availability of +the device. If not available, simply increment the device index and +returns. dev will be incremented also later (`step 7 +<7) Set the PCI driver data and return zero._>`__). + +2) Create a card instance +~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + struct snd_card *card; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + + +The details will be explained in the section `Management of Cards and +Components`_. + +3) Create a main component +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In this part, the PCI resources are allocated. + +:: + + struct mychip *chip; + .... + err = snd_mychip_create(card, pci, &chip); + if (err < 0) + goto error; + +The details will be explained in the section `PCI Resource +Management`_. + +When something goes wrong, the probe function needs to deal with the +error. In this example, we have a single error handling path placed +at the end of the function. + +:: + + error: + snd_card_free(card); + return err; + +Since each component can be properly freed, the single +:c:func:`snd_card_free()` call should suffice in most cases. + + +4) Set the driver ID and name strings. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->port, chip->irq); + +The driver field holds the minimal ID string of the chip. This is used +by alsa-lib's configurator, so keep it simple but unique. Even the +same driver can have different driver IDs to distinguish the +functionality of each chip type. + +The shortname field is a string shown as more verbose name. The longname +field contains the information shown in ``/proc/asound/cards``. + +5) Create other components, such as mixer, MIDI, etc. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +Here you define the basic components such as `PCM <PCM Interface_>`__, +mixer (e.g. `AC97 <API for AC97 Codec_>`__), MIDI (e.g. +`MPU-401 <MIDI (MPU401-UART) Interface_>`__), and other interfaces. +Also, if you want a `proc file <Proc Interface_>`__, define it here, +too. + +6) Register the card instance. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + err = snd_card_register(card); + if (err < 0) + goto error; + +Will be explained in the section `Management of Cards and +Components`_, too. + +7) Set the PCI driver data and return zero. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + pci_set_drvdata(pci, card); + dev++; + return 0; + +In the above, the card record is stored. This pointer is used in the +remove callback and power-management callbacks, too. + +Destructor +---------- + +The destructor, remove callback, simply releases the card instance. Then +the ALSA middle layer will release all the attached components +automatically. + +It would be typically just calling :c:func:`snd_card_free()`: + +:: + + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + } + + +The above code assumes that the card pointer is set to the PCI driver +data. + +Header Files +------------ + +For the above example, at least the following include files are +necessary. + +:: + + #include <linux/init.h> + #include <linux/pci.h> + #include <linux/slab.h> + #include <sound/core.h> + #include <sound/initval.h> + +where the last one is necessary only when module options are defined +in the source file. If the code is split into several files, the files +without module options don't need them. + +In addition to these headers, you'll need ``<linux/interrupt.h>`` for +interrupt handling, and ``<linux/io.h>`` for I/O access. If you use the +:c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need +to include ``<linux/delay.h>`` too. + +The ALSA interfaces like the PCM and control APIs are defined in other +``<sound/xxx.h>`` header files. They have to be included after +``<sound/core.h>``. + +Management of Cards and Components +================================== + +Card Instance +------------- + +For each soundcard, a “card” record must be allocated. + +A card record is the headquarters of the soundcard. It manages the whole +list of devices (components) on the soundcard, such as PCM, mixers, +MIDI, synthesizer, and so on. Also, the card record holds the ID and the +name strings of the card, manages the root of proc files, and controls +the power-management states and hotplug disconnections. The component +list on the card record is used to manage the correct release of +resources at destruction. + +As mentioned above, to create a card instance, call +:c:func:`snd_card_new()`. + +:: + + struct snd_card *card; + int err; + err = snd_card_new(&pci->dev, index, id, module, extra_size, &card); + + +The function takes six arguments: the parent device pointer, the +card-index number, the id string, the module pointer (usually +``THIS_MODULE``), the size of extra-data space, and the pointer to +return the card instance. The extra_size argument is used to allocate +card->private_data for the chip-specific data. Note that these data are +allocated by :c:func:`snd_card_new()`. + +The first argument, the pointer of struct device, specifies the parent +device. For PCI devices, typically ``&pci->`` is passed there. + +Components +---------- + +After the card is created, you can attach the components (devices) to +the card instance. In an ALSA driver, a component is represented as a +struct snd_device object. A component +can be a PCM instance, a control interface, a raw MIDI interface, etc. +Each such instance has one component entry. + +A component can be created via :c:func:`snd_device_new()` +function. + +:: + + snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); + +This takes the card pointer, the device-level (``SNDRV_DEV_XXX``), the +data pointer, and the callback pointers (``&ops``). The device-level +defines the type of components and the order of registration and +de-registration. For most components, the device-level is already +defined. For a user-defined component, you can use +``SNDRV_DEV_LOWLEVEL``. + +This function itself doesn't allocate the data space. The data must be +allocated manually beforehand, and its pointer is passed as the +argument. This pointer (``chip`` in the above example) is used as the +identifier for the instance. + +Each pre-defined ALSA component such as ac97 and pcm calls +:c:func:`snd_device_new()` inside its constructor. The destructor +for each component is defined in the callback pointers. Hence, you don't +need to take care of calling a destructor for such a component. + +If you wish to create your own component, you need to set the destructor +function to the dev_free callback in the ``ops``, so that it can be +released automatically via :c:func:`snd_card_free()`. The next +example will show an implementation of chip-specific data. + +Chip-Specific Data +------------------ + +Chip-specific information, e.g. the I/O port address, its resource +pointer, or the irq number, is stored in the chip-specific record. + +:: + + struct mychip { + .... + }; + + +In general, there are two ways of allocating the chip record. + +1. Allocating via :c:func:`snd_card_new()`. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +As mentioned above, you can pass the extra-data-length to the 5th +argument of :c:func:`snd_card_new()`, i.e. + +:: + + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + +struct mychip is the type of the chip record. + +In return, the allocated record can be accessed as + +:: + + struct mychip *chip = card->private_data; + +With this method, you don't have to allocate twice. The record is +released together with the card instance. + +2. Allocating an extra device. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +After allocating a card instance via :c:func:`snd_card_new()` +(with ``0`` on the 4th arg), call :c:func:`kzalloc()`. + +:: + + struct snd_card *card; + struct mychip *chip; + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + ..... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + +The chip record should have the field to hold the card pointer at least, + +:: + + struct mychip { + struct snd_card *card; + .... + }; + + +Then, set the card pointer in the returned chip instance. + +:: + + chip->card = card; + +Next, initialize the fields, and register this chip record as a +low-level device with a specified ``ops``, + +:: + + static const struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + .... + snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + +:c:func:`snd_mychip_dev_free()` is the device-destructor +function, which will call the real destructor. + +:: + + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + +where :c:func:`snd_mychip_free()` is the real destructor. + +The demerit of this method is the obviously more amount of codes. +The merit is, however, you can trigger the own callback at registering +and disconnecting the card via setting in snd_device_ops. +About the registering and disconnecting the card, see the subsections +below. + + +Registration and Release +------------------------ + +After all components are assigned, register the card instance by calling +:c:func:`snd_card_register()`. Access to the device files is +enabled at this point. That is, before +:c:func:`snd_card_register()` is called, the components are safely +inaccessible from external side. If this call fails, exit the probe +function after releasing the card via :c:func:`snd_card_free()`. + +For releasing the card instance, you can call simply +:c:func:`snd_card_free()`. As mentioned earlier, all components +are released automatically by this call. + +For a device which allows hotplugging, you can use +:c:func:`snd_card_free_when_closed()`. This one will postpone +the destruction until all devices are closed. + +PCI Resource Management +======================= + +Full Code Example +----------------- + +In this section, we'll complete the chip-specific constructor, +destructor and PCI entries. Example code is shown first, below. + +:: + + struct mychip { + struct snd_card *card; + struct pci_dev *pci; + + unsigned long port; + int irq; + }; + + static int snd_mychip_free(struct mychip *chip) + { + /* disable hardware here if any */ + .... /* (not implemented in this document) */ + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, chip); + /* release the I/O ports & memory */ + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; + } + + /* chip-specific constructor */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static const struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + card->sync_irq = chip->irq; + + /* (2) initialization of the chip hardware */ + .... /* (not implemented in this document) */ + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* PCI IDs */ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + + /* pci_driver definition */ + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + + /* module initialization */ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + /* module clean up */ + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + + EXPORT_NO_SYMBOLS; /* for old kernels only */ + +Some Hafta's +------------ + +The allocation of PCI resources is done in the ``probe`` function, and +usually an extra :c:func:`xxx_create()` function is written for this +purpose. + +In the case of PCI devices, you first have to call the +:c:func:`pci_enable_device()` function before allocating +resources. Also, you need to set the proper PCI DMA mask to limit the +accessed I/O range. In some cases, you might need to call +:c:func:`pci_set_master()` function, too. + +Suppose the 28bit mask, and the code to be added would be like: + +:: + + err = pci_enable_device(pci); + if (err < 0) + return err; + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + +Resource Allocation +------------------- + +The allocation of I/O ports and irqs is done via standard kernel +functions. These resources must be released in the destructor +function (see below). + +Now assume that the PCI device has an I/O port with 8 bytes and an +interrupt. Then struct mychip will have the +following fields: + +:: + + struct mychip { + struct snd_card *card; + + unsigned long port; + int irq; + }; + + +For an I/O port (and also a memory region), you need to have the +resource pointer for the standard resource management. For an irq, you +have to keep only the irq number (integer). But you need to initialize +this number as -1 before actual allocation, since irq 0 is valid. The +port address and its resource pointer can be initialized as null by +:c:func:`kzalloc()` automatically, so you don't have to take care of +resetting them. + +The allocation of an I/O port is done like this: + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + +It will reserve the I/O port region of 8 bytes of the given PCI device. +The returned value, ``chip->res_port``, is allocated via +:c:func:`kmalloc()` by :c:func:`request_region()`. The pointer +must be released via :c:func:`kfree()`, but there is a problem with +this. This issue will be explained later. + +The allocation of an interrupt source is done like this: + +:: + + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + +where :c:func:`snd_mychip_interrupt()` is the interrupt handler +defined `later <PCM Interrupt Handler_>`__. Note that +``chip->irq`` should be defined only when :c:func:`request_irq()` +succeeded. + +On the PCI bus, interrupts can be shared. Thus, ``IRQF_SHARED`` is used +as the interrupt flag of :c:func:`request_irq()`. + +The last argument of :c:func:`request_irq()` is the data pointer +passed to the interrupt handler. Usually, the chip-specific record is +used for that, but you can use what you like, too. + +I won't give details about the interrupt handler at this point, but at +least its appearance can be explained now. The interrupt handler looks +usually like the following: + +:: + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + .... + return IRQ_HANDLED; + } + +After requesting the IRQ, you can passed it to ``card->sync_irq`` +field: +:: + + card->irq = chip->irq; + +This allows PCM core automatically performing +:c:func:`synchronize_irq()` at the necessary timing like ``hw_free``. +See the later section `sync_stop callback`_ for details. + +Now let's write the corresponding destructor for the resources above. +The role of destructor is simple: disable the hardware (if already +activated) and release the resources. So far, we have no hardware part, +so the disabling code is not written here. + +To release the resources, the “check-and-release” method is a safer way. +For the interrupt, do like this: + +:: + + if (chip->irq >= 0) + free_irq(chip->irq, chip); + +Since the irq number can start from 0, you should initialize +``chip->irq`` with a negative value (e.g. -1), so that you can check +the validity of the irq number as above. + +When you requested I/O ports or memory regions via +:c:func:`pci_request_region()` or +:c:func:`pci_request_regions()` like in this example, release the +resource(s) using the corresponding function, +:c:func:`pci_release_region()` or +:c:func:`pci_release_regions()`. + +:: + + pci_release_regions(chip->pci); + +When you requested manually via :c:func:`request_region()` or +:c:func:`request_mem_region()`, you can release it via +:c:func:`release_resource()`. Suppose that you keep the resource +pointer returned from :c:func:`request_region()` in +chip->res_port, the release procedure looks like: + +:: + + release_and_free_resource(chip->res_port); + +Don't forget to call :c:func:`pci_disable_device()` before the +end. + +And finally, release the chip-specific record. + +:: + + kfree(chip); + +We didn't implement the hardware disabling part in the above. If you +need to do this, please note that the destructor may be called even +before the initialization of the chip is completed. It would be better +to have a flag to skip hardware disabling if the hardware was not +initialized yet. + +When the chip-data is assigned to the card using +:c:func:`snd_device_new()` with ``SNDRV_DEV_LOWLELVEL`` , its +destructor is called at the last. That is, it is assured that all other +components like PCMs and controls have already been released. You don't +have to stop PCMs, etc. explicitly, but just call low-level hardware +stopping. + +The management of a memory-mapped region is almost as same as the +management of an I/O port. You'll need three fields like the +following: + +:: + + struct mychip { + .... + unsigned long iobase_phys; + void __iomem *iobase_virt; + }; + +and the allocation would be like below: + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap(chip->iobase_phys, + pci_resource_len(pci, 0)); + +and the corresponding destructor would be: + +:: + + static int snd_mychip_free(struct mychip *chip) + { + .... + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } + +Of course, a modern way with :c:func:`pci_iomap()` will make things a +bit easier, too. + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + return err; + } + chip->iobase_virt = pci_iomap(pci, 0, 0); + +which is paired with :c:func:`pci_iounmap()` at destructor. + + +PCI Entries +----------- + +So far, so good. Let's finish the missing PCI stuff. At first, we need a +struct pci_device_id table for +this chipset. It's a table of PCI vendor/device ID number, and some +masks. + +For example, + +:: + + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + +The first and second fields of the struct pci_device_id are the vendor +and device IDs. If you have no reason to filter the matching devices, you can +leave the remaining fields as above. The last field of the +struct pci_device_id contains private data for this entry. You can specify +any value here, for example, to define specific operations for supported +device IDs. Such an example is found in the intel8x0 driver. + +The last entry of this list is the terminator. You must specify this +all-zero entry. + +Then, prepare the struct pci_driver +record: + +:: + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + +The ``probe`` and ``remove`` functions have already been defined in +the previous sections. The ``name`` field is the name string of this +device. Note that you must not use a slash “/” in this string. + +And at last, the module entries: + +:: + + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + +Note that these module entries are tagged with ``__init`` and ``__exit`` +prefixes. + +That's all! + +PCM Interface +============= + +General +------- + +The PCM middle layer of ALSA is quite powerful and it is only necessary +for each driver to implement the low-level functions to access its +hardware. + +For accessing to the PCM layer, you need to include ``<sound/pcm.h>`` +first. In addition, ``<sound/pcm_params.h>`` might be needed if you +access to some functions related with hw_param. + +Each card device can have up to four pcm instances. A pcm instance +corresponds to a pcm device file. The limitation of number of instances +comes only from the available bit size of the Linux's device numbers. +Once when 64bit device number is used, we'll have more pcm instances +available. + +A pcm instance consists of pcm playback and capture streams, and each +pcm stream consists of one or more pcm substreams. Some soundcards +support multiple playback functions. For example, emu10k1 has a PCM +playback of 32 stereo substreams. In this case, at each open, a free +substream is (usually) automatically chosen and opened. Meanwhile, when +only one substream exists and it was already opened, the successful open +will either block or error with ``EAGAIN`` according to the file open +mode. But you don't have to care about such details in your driver. The +PCM middle layer will take care of such work. + +Full Code Example +----------------- + +The example code below does not include any hardware access routines but +shows only the skeleton, how to build up the PCM interfaces. + +:: + + #include <sound/pcm.h> + .... + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* open callback */ + static int snd_mychip_playback_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_playback_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_capture_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_capture_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + } + + /* hw_params callback */ + static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) + { + /* the hardware-specific codes will be here */ + .... + return 0; + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + /* the hardware-specific codes will be here */ + .... + return 0; + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pci->dev, + 64*1024, 64*1024); + return 0; + } + + +PCM Constructor +--------------- + +A pcm instance is allocated by the :c:func:`snd_pcm_new()` +function. It would be better to create a constructor for pcm, namely, + +:: + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } + +The :c:func:`snd_pcm_new()` function takes four arguments. The +first argument is the card pointer to which this pcm is assigned, and +the second is the ID string. + +The third argument (``index``, 0 in the above) is the index of this new +pcm. It begins from zero. If you create more than one pcm instances, +specify the different numbers in this argument. For example, ``index = +1`` for the second PCM device. + +The fourth and fifth arguments are the number of substreams for playback +and capture, respectively. Here 1 is used for both arguments. When no +playback or capture substreams are available, pass 0 to the +corresponding argument. + +If a chip supports multiple playbacks or captures, you can specify more +numbers, but they must be handled properly in open/close, etc. +callbacks. When you need to know which substream you are referring to, +then it can be obtained from struct snd_pcm_substream data passed to each +callback as follows: + +:: + + struct snd_pcm_substream *substream; + int index = substream->number; + + +After the pcm is created, you need to set operators for each pcm stream. + +:: + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + +The operators are defined typically like this: + +:: + + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_pcm_open, + .close = snd_mychip_pcm_close, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + +All the callbacks are described in the Operators_ subsection. + +After setting the operators, you probably will want to pre-allocate the +buffer and set up the managed allocation mode. +For that, simply call the following: + +:: + + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pci->dev, + 64*1024, 64*1024); + +It will allocate a buffer up to 64kB as default. Buffer management +details will be described in the later section `Buffer and Memory +Management`_. + +Additionally, you can set some extra information for this pcm in +``pcm->info_flags``. The available values are defined as +``SNDRV_PCM_INFO_XXX`` in ``<sound/asound.h>``, which is used for the +hardware definition (described later). When your soundchip supports only +half-duplex, specify like this: + +:: + + pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + + +... And the Destructor? +----------------------- + +The destructor for a pcm instance is not always necessary. Since the pcm +device will be released by the middle layer code automatically, you +don't have to call the destructor explicitly. + +The destructor would be necessary if you created special records +internally and needed to release them. In such a case, set the +destructor function to ``pcm->private_free``: + +:: + + static void mychip_pcm_free(struct snd_pcm *pcm) + { + struct mychip *chip = snd_pcm_chip(pcm); + /* free your own data */ + kfree(chip->my_private_pcm_data); + /* do what you like else */ + .... + } + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } + + + +Runtime Pointer - The Chest of PCM Information +---------------------------------------------- + +When the PCM substream is opened, a PCM runtime instance is allocated +and assigned to the substream. This pointer is accessible via +``substream->runtime``. This runtime pointer holds most information you +need to control the PCM: the copy of hw_params and sw_params +configurations, the buffer pointers, mmap records, spinlocks, etc. + +The definition of runtime instance is found in ``<sound/pcm.h>``. Here +are the contents of this file: + +:: + + struct _snd_pcm_runtime { + /* -- Status -- */ + struct snd_pcm_substream *trigger_master; + snd_timestamp_t trigger_tstamp; /* trigger timestamp */ + int overrange; + snd_pcm_uframes_t avail_max; + snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + + /* -- HW params -- */ + snd_pcm_access_t access; /* access mode */ + snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ + snd_pcm_subformat_t subformat; /* subformat */ + unsigned int rate; /* rate in Hz */ + unsigned int channels; /* channels */ + snd_pcm_uframes_t period_size; /* period size */ + unsigned int periods; /* periods */ + snd_pcm_uframes_t buffer_size; /* buffer size */ + unsigned int tick_time; /* tick time */ + snd_pcm_uframes_t min_align; /* Min alignment for the format */ + size_t byte_align; + unsigned int frame_bits; + unsigned int sample_bits; + unsigned int info; + unsigned int rate_num; + unsigned int rate_den; + + /* -- SW params -- */ + struct timespec tstamp_mode; /* mmap timestamp is updated */ + unsigned int period_step; + unsigned int sleep_min; /* min ticks to sleep */ + snd_pcm_uframes_t start_threshold; + snd_pcm_uframes_t stop_threshold; + snd_pcm_uframes_t silence_threshold; /* Silence filling happens when + noise is nearest than this */ + snd_pcm_uframes_t silence_size; /* Silence filling size */ + snd_pcm_uframes_t boundary; /* pointers wrap point */ + + snd_pcm_uframes_t silenced_start; + snd_pcm_uframes_t silenced_size; + + snd_pcm_sync_id_t sync; /* hardware synchronization ID */ + + /* -- mmap -- */ + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + atomic_t mmap_count; + + /* -- locking / scheduling -- */ + spinlock_t lock; + wait_queue_head_t sleep; + struct timer_list tick_timer; + struct fasync_struct *fasync; + + /* -- private section -- */ + void *private_data; + void (*private_free)(struct snd_pcm_runtime *runtime); + + /* -- hardware description -- */ + struct snd_pcm_hardware hw; + struct snd_pcm_hw_constraints hw_constraints; + + /* -- timer -- */ + unsigned int timer_resolution; /* timer resolution */ + + /* -- DMA -- */ + unsigned char *dma_area; /* DMA area */ + dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ + size_t dma_bytes; /* size of DMA area */ + + struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) + /* -- OSS things -- */ + struct snd_pcm_oss_runtime oss; + #endif + }; + + +For the operators (callbacks) of each sound driver, most of these +records are supposed to be read-only. Only the PCM middle-layer changes +/ updates them. The exceptions are the hardware description (hw) DMA +buffer information and the private data. Besides, if you use the +standard managed buffer allocation mode, you don't need to set the +DMA buffer information by yourself. + +In the sections below, important records are explained. + +Hardware Description +~~~~~~~~~~~~~~~~~~~~ + +The hardware descriptor (struct snd_pcm_hardware) contains the definitions of +the fundamental hardware configuration. Above all, you'll need to define this +in the `PCM open callback`_. Note that the runtime instance holds the copy of +the descriptor, not the pointer to the existing descriptor. That is, +in the open callback, you can modify the copied descriptor +(``runtime->hw``) as you need. For example, if the maximum number of +channels is 1 only on some chip models, you can still use the same +hardware descriptor and change the channels_max later: + +:: + + struct snd_pcm_runtime *runtime = substream->runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; + +Typically, you'll have a hardware descriptor as below: + +:: + + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + +- The ``info`` field contains the type and capabilities of this + pcm. The bit flags are defined in ``<sound/asound.h>`` as + ``SNDRV_PCM_INFO_XXX``. Here, at least, you have to specify whether + the mmap is supported and which interleaved format is + supported. When the hardware supports mmap, add the + ``SNDRV_PCM_INFO_MMAP`` flag here. When the hardware supports the + interleaved or the non-interleaved formats, + ``SNDRV_PCM_INFO_INTERLEAVED`` or ``SNDRV_PCM_INFO_NONINTERLEAVED`` + flag must be set, respectively. If both are supported, you can set + both, too. + + In the above example, ``MMAP_VALID`` and ``BLOCK_TRANSFER`` are + specified for the OSS mmap mode. Usually both are set. Of course, + ``MMAP_VALID`` is set only if the mmap is really supported. + + The other possible flags are ``SNDRV_PCM_INFO_PAUSE`` and + ``SNDRV_PCM_INFO_RESUME``. The ``PAUSE`` bit means that the pcm + supports the “pause” operation, while the ``RESUME`` bit means that + the pcm supports the full “suspend/resume” operation. If the + ``PAUSE`` flag is set, the ``trigger`` callback below must handle + the corresponding (pause push/release) commands. The suspend/resume + trigger commands can be defined even without the ``RESUME`` + flag. See `Power Management`_ section for details. + + When the PCM substreams can be synchronized (typically, + synchronized start/stop of a playback and a capture streams), you + can give ``SNDRV_PCM_INFO_SYNC_START``, too. In this case, you'll + need to check the linked-list of PCM substreams in the trigger + callback. This will be described in the later section. + +- ``formats`` field contains the bit-flags of supported formats + (``SNDRV_PCM_FMTBIT_XXX``). If the hardware supports more than one + format, give all or'ed bits. In the example above, the signed 16bit + little-endian format is specified. + +- ``rates`` field contains the bit-flags of supported rates + (``SNDRV_PCM_RATE_XXX``). When the chip supports continuous rates, + pass ``CONTINUOUS`` bit additionally. The pre-defined rate bits are + provided only for typical rates. If your chip supports + unconventional rates, you need to add the ``KNOT`` bit and set up + the hardware constraint manually (explained later). + +- ``rate_min`` and ``rate_max`` define the minimum and maximum sample + rate. This should correspond somehow to ``rates`` bits. + +- ``channel_min`` and ``channel_max`` define, as you might already + expected, the minimum and maximum number of channels. + +- ``buffer_bytes_max`` defines the maximum buffer size in + bytes. There is no ``buffer_bytes_min`` field, since it can be + calculated from the minimum period size and the minimum number of + periods. Meanwhile, ``period_bytes_min`` and define the minimum and + maximum size of the period in bytes. ``periods_max`` and + ``periods_min`` define the maximum and minimum number of periods in + the buffer. + + The “period” is a term that corresponds to a fragment in the OSS + world. The period defines the size at which a PCM interrupt is + generated. This size strongly depends on the hardware. Generally, + the smaller period size will give you more interrupts, that is, + more controls. In the case of capture, this size defines the input + latency. On the other hand, the whole buffer size defines the + output latency for the playback direction. + +- There is also a field ``fifo_size``. This specifies the size of the + hardware FIFO, but currently it is neither used in the driver nor + in the alsa-lib. So, you can ignore this field. + +PCM Configurations +~~~~~~~~~~~~~~~~~~ + +Ok, let's go back again to the PCM runtime records. The most +frequently referred records in the runtime instance are the PCM +configurations. The PCM configurations are stored in the runtime +instance after the application sends ``hw_params`` data via +alsa-lib. There are many fields copied from hw_params and sw_params +structs. For example, ``format`` holds the format type chosen by the +application. This field contains the enum value +``SNDRV_PCM_FORMAT_XXX``. + +One thing to be noted is that the configured buffer and period sizes +are stored in “frames” in the runtime. In the ALSA world, ``1 frame = +channels \* samples-size``. For conversion between frames and bytes, +you can use the :c:func:`frames_to_bytes()` and +:c:func:`bytes_to_frames()` helper functions. + +:: + + period_bytes = frames_to_bytes(runtime, runtime->period_size); + +Also, many software parameters (sw_params) are stored in frames, too. +Please check the type of the field. ``snd_pcm_uframes_t`` is for the +frames as unsigned integer while ``snd_pcm_sframes_t`` is for the +frames as signed integer. + +DMA Buffer Information +~~~~~~~~~~~~~~~~~~~~~~ + +The DMA buffer is defined by the following four fields, ``dma_area``, +``dma_addr``, ``dma_bytes`` and ``dma_private``. The ``dma_area`` +holds the buffer pointer (the logical address). You can call +:c:func:`memcpy()` from/to this pointer. Meanwhile, ``dma_addr`` holds +the physical address of the buffer. This field is specified only when +the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer +in bytes. ``dma_private`` is used for the ALSA DMA allocator. + +If you use either the managed buffer allocation mode or the standard +API function :c:func:`snd_pcm_lib_malloc_pages()` for allocating the buffer, +these fields are set by the ALSA middle layer, and you should *not* +change them by yourself. You can read them but not write them. On the +other hand, if you want to allocate the buffer by yourself, you'll +need to manage it in hw_params callback. At least, ``dma_bytes`` is +mandatory. ``dma_area`` is necessary when the buffer is mmapped. If +your driver doesn't support mmap, this field is not +necessary. ``dma_addr`` is also optional. You can use dma_private as +you like, too. + +Running Status +~~~~~~~~~~~~~~ + +The running status can be referred via ``runtime->status``. This is +the pointer to the struct snd_pcm_mmap_status record. +For example, you can get the current +DMA hardware pointer via ``runtime->status->hw_ptr``. + +The DMA application pointer can be referred via ``runtime->control``, +which points to the struct snd_pcm_mmap_control record. +However, accessing directly to this value is not recommended. + +Private Data +~~~~~~~~~~~~ + +You can allocate a record for the substream and store it in +``runtime->private_data``. Usually, this is done in the `PCM open +callback`_. Don't mix this with ``pcm->private_data``. The +``pcm->private_data`` usually points to the chip instance assigned +statically at the creation of PCM, while the ``runtime->private_data`` +points to a dynamic data structure created at the PCM open +callback. + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct my_pcm_data *data; + .... + data = kmalloc(sizeof(*data), GFP_KERNEL); + substream->runtime->private_data = data; + .... + } + + +The allocated object must be released in the `close callback`_. + +Operators +--------- + +OK, now let me give details about each pcm callback (``ops``). In +general, every callback must return 0 if successful, or a negative +error number such as ``-EINVAL``. To choose an appropriate error +number, it is advised to check what value other parts of the kernel +return when the same kind of request fails. + +The callback function takes at least the argument with +struct snd_pcm_substream pointer. To retrieve the chip +record from the given substream instance, you can use the following +macro. + +:: + + int xxx() { + struct mychip *chip = snd_pcm_substream_chip(substream); + .... + } + +The macro reads ``substream->private_data``, which is a copy of +``pcm->private_data``. You can override the former if you need to +assign different data records per PCM substream. For example, the +cmi8330 driver assigns different ``private_data`` for playback and +capture directions, because it uses two different codecs (SB- and +AD-compatible) for different directions. + +PCM open callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream); + +This is called when a pcm substream is opened. + +At least, here you have to initialize the ``runtime->hw`` +record. Typically, this is done by like this: + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } + +where ``snd_mychip_playback_hw`` is the pre-defined hardware +description. + +You can allocate a private data in this callback, as described in +`Private Data`_ section. + +If the hardware configuration needs more constraints, set the hardware +constraints here, too. See Constraints_ for more details. + +close callback +~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream); + + +Obviously, this is called when a pcm substream is closed. + +Any private instance for a pcm substream allocated in the ``open`` +callback will be released here. + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream) + { + .... + kfree(substream->runtime->private_data); + .... + } + +ioctl callback +~~~~~~~~~~~~~~ + +This is used for any special call to pcm ioctls. But usually you can +leave it as NULL, then PCM core calls the generic ioctl callback +function :c:func:`snd_pcm_lib_ioctl()`. If you need to deal with the +unique setup of channel info or reset procedure, you can pass your own +callback function here. + +hw_params callback +~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); + +This is called when the hardware parameter (``hw_params``) is set up +by the application, that is, once when the buffer size, the period +size, the format, etc. are defined for the pcm substream. + +Many hardware setups should be done in this callback, including the +allocation of buffers. + +Parameters to be initialized are retrieved by +:c:func:`params_xxx()` macros. + +When you set up the managed buffer allocation mode for the substream, +a buffer is already allocated before this callback gets +called. Alternatively, you can call a helper function below for +allocating the buffer, too. + +:: + + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + +:c:func:`snd_pcm_lib_malloc_pages()` is available only when the +DMA buffers have been pre-allocated. See the section `Buffer Types`_ +for more details. + +Note that this and ``prepare`` callbacks may be called multiple times +per initialization. For example, the OSS emulation may call these +callbacks at each change via its ioctl. + +Thus, you need to be careful not to allocate the same buffers many +times, which will lead to memory leaks! Calling the helper function +above many times is OK. It will release the previous buffer +automatically when it was already allocated. + +Another note is that this callback is non-atomic (schedulable) as +default, i.e. when no ``nonatomic`` flag set. This is important, +because the ``trigger`` callback is atomic (non-schedulable). That is, +mutexes or any schedule-related functions are not available in +``trigger`` callback. Please see the subsection Atomicity_ for +details. + +hw_free callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_free(struct snd_pcm_substream *substream); + +This is called to release the resources allocated via +``hw_params``. + +This function is always called before the close callback is called. +Also, the callback may be called multiple times, too. Keep track +whether the resource was already released. + +When you have set up the managed buffer allocation mode for the PCM +substream, the allocated PCM buffer will be automatically released +after this callback gets called. Otherwise you'll have to release the +buffer manually. Typically, when the buffer was allocated from the +pre-allocated pool, you can use the standard API function +:c:func:`snd_pcm_lib_malloc_pages()` like: + +:: + + snd_pcm_lib_free_pages(substream); + +prepare callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_prepare(struct snd_pcm_substream *substream); + +This callback is called when the pcm is “prepared”. You can set the +format type, sample rate, etc. here. The difference from ``hw_params`` +is that the ``prepare`` callback will be called each time +:c:func:`snd_pcm_prepare()` is called, i.e. when recovering after +underruns, etc. + +Note that this callback is now non-atomic. You can use +schedule-related functions safely in this callback. + +In this and the following callbacks, you can refer to the values via +the runtime record, ``substream->runtime``. For example, to get the +current rate, format or channels, access to ``runtime->rate``, +``runtime->format`` or ``runtime->channels``, respectively. The +physical address of the allocated buffer is set to +``runtime->dma_area``. The buffer and period sizes are in +``runtime->buffer_size`` and ``runtime->period_size``, respectively. + +Be careful that this callback will be called many times at each setup, +too. + +trigger callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); + +This is called when the pcm is started, stopped or paused. + +Which action is specified in the second argument, +``SNDRV_PCM_TRIGGER_XXX`` in ``<sound/pcm.h>``. At least, the ``START`` +and ``STOP`` commands must be defined in this callback. + +:: + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + break; + default: + return -EINVAL; + } + +When the pcm supports the pause operation (given in the info field of +the hardware table), the ``PAUSE_PUSH`` and ``PAUSE_RELEASE`` commands +must be handled here, too. The former is the command to pause the pcm, +and the latter to restart the pcm again. + +When the pcm supports the suspend/resume operation, regardless of full +or partial suspend/resume support, the ``SUSPEND`` and ``RESUME`` +commands must be handled, too. These commands are issued when the +power-management status is changed. Obviously, the ``SUSPEND`` and +``RESUME`` commands suspend and resume the pcm substream, and usually, +they are identical to the ``STOP`` and ``START`` commands, respectively. +See the `Power Management`_ section for details. + +As mentioned, this callback is atomic as default unless ``nonatomic`` +flag set, and you cannot call functions which may sleep. The +``trigger`` callback should be as minimal as possible, just really +triggering the DMA. The other stuff should be initialized +``hw_params`` and ``prepare`` callbacks properly beforehand. + +sync_stop callback +~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_sync_stop(struct snd_pcm_substream *substream); + +This callback is optional, and NULL can be passed. It's called after +the PCM core stops the stream and changes the stream state +``prepare``, ``hw_params`` or ``hw_free``. +Since the IRQ handler might be still pending, we need to wait until +the pending task finishes before moving to the next step; otherwise it +might lead to a crash due to resource conflicts or access to the freed +resources. A typical behavior is to call a synchronization function +like :c:func:`synchronize_irq()` here. + +For majority of drivers that need only a call of +:c:func:`synchronize_irq()`, there is a simpler setup, too. +While keeping NULL to ``sync_stop`` PCM callback, the driver can set +``card->sync_irq`` field to store the valid interrupt number after +requesting an IRQ, instead. Then PCM core will look call +:c:func:`synchronize_irq()` with the given IRQ appropriately. + +If the IRQ handler is released at the card destructor, you don't need +to clear ``card->sync_irq``, as the card itself is being released. +So, usually you'll need to add just a single line for assigning +``card->sync_irq`` in the driver code unless the driver re-acquires +the IRQ. When the driver frees and re-acquires the IRQ dynamically +(e.g. for suspend/resume), it needs to clear and re-set +``card->sync_irq`` again appropriately. + +pointer callback +~~~~~~~~~~~~~~~~ + +:: + + static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) + +This callback is called when the PCM middle layer inquires the current +hardware position on the buffer. The position must be returned in +frames, ranging from 0 to ``buffer_size - 1``. + +This is called usually from the buffer-update routine in the pcm +middle layer, which is invoked when :c:func:`snd_pcm_period_elapsed()` +is called in the interrupt routine. Then the pcm middle layer updates +the position and calculates the available space, and wakes up the +sleeping poll threads, etc. + +This callback is also atomic as default. + +copy_user, copy_kernel and fill_silence ops +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +These callbacks are not mandatory, and can be omitted in most cases. +These callbacks are used when the hardware buffer cannot be in the +normal memory space. Some chips have their own buffer on the hardware +which is not mappable. In such a case, you have to transfer the data +manually from the memory buffer to the hardware buffer. Or, if the +buffer is non-contiguous on both physical and virtual memory spaces, +these callbacks must be defined, too. + +If these two callbacks are defined, copy and set-silence operations +are done by them. The detailed will be described in the later section +`Buffer and Memory Management`_. + +ack callback +~~~~~~~~~~~~ + +This callback is also not mandatory. This callback is called when the +``appl_ptr`` is updated in read or write operations. Some drivers like +emu10k1-fx and cs46xx need to track the current ``appl_ptr`` for the +internal buffer, and this callback is useful only for such a purpose. + +This callback is atomic as default. + +page callback +~~~~~~~~~~~~~ + +This callback is optional too. The mmap calls this callback to get the +page fault address. + +Since the recent changes, you need no special callback any longer for +the standard SG-buffer or vmalloc-buffer. Hence this callback should +be rarely used. + +mmap calllback +~~~~~~~~~~~~~~ + +This is another optional callback for controlling mmap behavior. +Once when defined, PCM core calls this callback when a page is +memory-mapped instead of dealing via the standard helper. +If you need special handling (due to some architecture or +device-specific issues), implement everything here as you like. + + +PCM Interrupt Handler +--------------------- + +The rest of pcm stuff is the PCM interrupt handler. The role of PCM +interrupt handler in the sound driver is to update the buffer position +and to tell the PCM middle layer when the buffer position goes across +the prescribed period size. To inform this, call the +:c:func:`snd_pcm_period_elapsed()` function. + +There are several types of sound chips to generate the interrupts. + +Interrupts at the period (fragment) boundary +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This is the most frequently found type: the hardware generates an +interrupt at each period boundary. In this case, you can call +:c:func:`snd_pcm_period_elapsed()` at each interrupt. + +:c:func:`snd_pcm_period_elapsed()` takes the substream pointer as +its argument. Thus, you need to keep the substream pointer accessible +from the chip instance. For example, define ``substream`` field in the +chip record to hold the current running substream pointer, and set the +pointer value at ``open`` callback (and reset at ``close`` callback). + +If you acquire a spinlock in the interrupt handler, and the lock is used +in other pcm callbacks, too, then you have to release the lock before +calling :c:func:`snd_pcm_period_elapsed()`, because +:c:func:`snd_pcm_period_elapsed()` calls other pcm callbacks +inside. + +Typical code would be like: + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + /* call updater, unlock before it */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(chip->substream); + spin_lock(&chip->lock); + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +High frequency timer interrupts +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This happens when the hardware doesn't generate interrupts at the period +boundary but issues timer interrupts at a fixed timer rate (e.g. es1968 +or ymfpci drivers). In this case, you need to check the current hardware +position and accumulate the processed sample length at each interrupt. +When the accumulated size exceeds the period size, call +:c:func:`snd_pcm_period_elapsed()` and reset the accumulator. + +Typical code would be like the following. + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + unsigned int last_ptr, size; + /* get the current hardware pointer (in frames) */ + last_ptr = get_hw_ptr(chip); + /* calculate the processed frames since the + * last update + */ + if (last_ptr < chip->last_ptr) + size = runtime->buffer_size + last_ptr + - chip->last_ptr; + else + size = last_ptr - chip->last_ptr; + /* remember the last updated point */ + chip->last_ptr = last_ptr; + /* accumulate the size */ + chip->size += size; + /* over the period boundary? */ + if (chip->size >= runtime->period_size) { + /* reset the accumulator */ + chip->size %= runtime->period_size; + /* call updater */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(substream); + spin_lock(&chip->lock); + } + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +On calling :c:func:`snd_pcm_period_elapsed()` +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In both cases, even if more than one period are elapsed, you don't have +to call :c:func:`snd_pcm_period_elapsed()` many times. Call only +once. And the pcm layer will check the current hardware pointer and +update to the latest status. + +Atomicity +--------- + +One of the most important (and thus difficult to debug) problems in +kernel programming are race conditions. In the Linux kernel, they are +usually avoided via spin-locks, mutexes or semaphores. In general, if a +race condition can happen in an interrupt handler, it has to be managed +atomically, and you have to use a spinlock to protect the critical +session. If the critical section is not in interrupt handler code and if +taking a relatively long time to execute is acceptable, you should use +mutexes or semaphores instead. + +As already seen, some pcm callbacks are atomic and some are not. For +example, the ``hw_params`` callback is non-atomic, while ``trigger`` +callback is atomic. This means, the latter is called already in a +spinlock held by the PCM middle layer. Please take this atomicity into +account when you choose a locking scheme in the callbacks. + +In the atomic callbacks, you cannot use functions which may call +:c:func:`schedule()` or go to :c:func:`sleep()`. Semaphores and +mutexes can sleep, and hence they cannot be used inside the atomic +callbacks (e.g. ``trigger`` callback). To implement some delay in such a +callback, please use :c:func:`udelay()` or :c:func:`mdelay()`. + +All three atomic callbacks (trigger, pointer, and ack) are called with +local interrupts disabled. + +The recent changes in PCM core code, however, allow all PCM operations +to be non-atomic. This assumes that the all caller sides are in +non-atomic contexts. For example, the function +:c:func:`snd_pcm_period_elapsed()` is called typically from the +interrupt handler. But, if you set up the driver to use a threaded +interrupt handler, this call can be in non-atomic context, too. In such +a case, you can set ``nonatomic`` filed of struct snd_pcm object +after creating it. When this flag is set, mutex and rwsem are used internally +in the PCM core instead of spin and rwlocks, so that you can call all PCM +functions safely in a non-atomic +context. + +Constraints +----------- + +If your chip supports unconventional sample rates, or only the limited +samples, you need to set a constraint for the condition. + +For example, in order to restrict the sample rates in the some supported +values, use :c:func:`snd_pcm_hw_constraint_list()`. You need to +call this function in the open callback. + +:: + + static unsigned int rates[] = + {4000, 10000, 22050, 44100}; + static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + static int snd_mychip_pcm_open(struct snd_pcm_substream *substream) + { + int err; + .... + err = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } + + + +There are many different constraints. Look at ``sound/pcm.h`` for a +complete list. You can even define your own constraint rules. For +example, let's suppose my_chip can manage a substream of 1 channel if +and only if the format is ``S16_LE``, otherwise it supports any format +specified in struct snd_pcm_hardware> (or in any other +constraint_list). You can build a rule like this: + +:: + + static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval ch; + + snd_interval_any(&ch); + if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } + + +Then you need to call this function to add your rule: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + +The rule function is called when an application sets the PCM format, and +it refines the number of channels accordingly. But an application may +set the number of channels before setting the format. Thus you also need +to define the inverse rule: + +:: + + static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask fmt; + + snd_mask_any(&fmt); /* Init the struct */ + if (c->min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } + + +... and in the open callback: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + +One typical usage of the hw constraints is to align the buffer size +with the period size. As default, ALSA PCM core doesn't enforce the +buffer size to be aligned with the period size. For example, it'd be +possible to have a combination like 256 period bytes with 999 buffer +bytes. + +Many device chips, however, require the buffer to be a multiple of +periods. In such a case, call +:c:func:`snd_pcm_hw_constraint_integer()` for +``SNDRV_PCM_HW_PARAM_PERIODS``. + +:: + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + +This assures that the number of periods is integer, hence the buffer +size is aligned with the period size. + +The hw constraint is a very much powerful mechanism to define the +preferred PCM configuration, and there are relevant helpers. +I won't give more details here, rather I would like to say, “Luke, use +the source.” + +Control Interface +================= + +General +------- + +The control interface is used widely for many switches, sliders, etc. +which are accessed from user-space. Its most important use is the mixer +interface. In other words, since ALSA 0.9.x, all the mixer stuff is +implemented on the control kernel API. + +ALSA has a well-defined AC97 control module. If your chip supports only +the AC97 and nothing else, you can skip this section. + +The control API is defined in ``<sound/control.h>``. Include this file +if you want to add your own controls. + +Definition of Controls +---------------------- + +To create a new control, you need to define the following three +callbacks: ``info``, ``get`` and ``put``. Then, define a +struct snd_kcontrol_new record, such as: + +:: + + + static struct snd_kcontrol_new my_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = my_control_info, + .get = my_control_get, + .put = my_control_put + }; + + +The ``iface`` field specifies the control type, +``SNDRV_CTL_ELEM_IFACE_XXX``, which is usually ``MIXER``. Use ``CARD`` +for global controls that are not logically part of the mixer. If the +control is closely associated with some specific device on the sound +card, use ``HWDEP``, ``PCM``, ``RAWMIDI``, ``TIMER``, or ``SEQUENCER``, +and specify the device number with the ``device`` and ``subdevice`` +fields. + +The ``name`` is the name identifier string. Since ALSA 0.9.x, the +control name is very important, because its role is classified from +its name. There are pre-defined standard control names. The details +are described in the `Control Names`_ subsection. + +The ``index`` field holds the index number of this control. If there +are several different controls with the same name, they can be +distinguished by the index number. This is the case when several +codecs exist on the card. If the index is zero, you can omit the +definition above. + +The ``access`` field contains the access type of this control. Give +the combination of bit masks, ``SNDRV_CTL_ELEM_ACCESS_XXX``, +there. The details will be explained in the `Access Flags`_ +subsection. + +The ``private_value`` field contains an arbitrary long integer value +for this record. When using the generic ``info``, ``get`` and ``put`` +callbacks, you can pass a value through this field. If several small +numbers are necessary, you can combine them in bitwise. Or, it's +possible to give a pointer (casted to unsigned long) of some record to +this field, too. + +The ``tlv`` field can be used to provide metadata about the control; +see the `Metadata`_ subsection. + +The other three are `Control Callbacks`_. + +Control Names +------------- + +There are some standards to define the control names. A control is +usually defined from the three parts as “SOURCE DIRECTION FUNCTION”. + +The first, ``SOURCE``, specifies the source of the control, and is a +string such as “Master”, “PCM”, “CD” and “Line”. There are many +pre-defined sources. + +The second, ``DIRECTION``, is one of the following strings according to +the direction of the control: “Playback”, “Capture”, “Bypass Playback” +and “Bypass Capture”. Or, it can be omitted, meaning both playback and +capture directions. + +The third, ``FUNCTION``, is one of the following strings according to +the function of the control: “Switch”, “Volume” and “Route”. + +The example of control names are, thus, “Master Capture Switch” or “PCM +Playback Volume”. + +There are some exceptions: + +Global capture and playback +~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +“Capture Source”, “Capture Switch” and “Capture Volume” are used for the +global capture (input) source, switch and volume. Similarly, “Playback +Switch” and “Playback Volume” are used for the global output gain switch +and volume. + +Tone-controls +~~~~~~~~~~~~~ + +tone-control switch and volumes are specified like “Tone Control - XXX”, +e.g. “Tone Control - Switch”, “Tone Control - Bass”, “Tone Control - +Center”. + +3D controls +~~~~~~~~~~~ + +3D-control switches and volumes are specified like “3D Control - XXX”, +e.g. “3D Control - Switch”, “3D Control - Center”, “3D Control - Space”. + +Mic boost +~~~~~~~~~ + +Mic-boost switch is set as “Mic Boost” or “Mic Boost (6dB)”. + +More precise information can be found in +``Documentation/sound/designs/control-names.rst``. + +Access Flags +------------ + +The access flag is the bitmask which specifies the access type of the +given control. The default access type is +``SNDRV_CTL_ELEM_ACCESS_READWRITE``, which means both read and write are +allowed to this control. When the access flag is omitted (i.e. = 0), it +is considered as ``READWRITE`` access as default. + +When the control is read-only, pass ``SNDRV_CTL_ELEM_ACCESS_READ`` +instead. In this case, you don't have to define the ``put`` callback. +Similarly, when the control is write-only (although it's a rare case), +you can use the ``WRITE`` flag instead, and you don't need the ``get`` +callback. + +If the control value changes frequently (e.g. the VU meter), +``VOLATILE`` flag should be given. This means that the control may be +changed without `Change notification`_. Applications should poll such +a control constantly. + +When the control is inactive, set the ``INACTIVE`` flag, too. There are +``LOCK`` and ``OWNER`` flags to change the write permissions. + +Control Callbacks +----------------- + +info callback +~~~~~~~~~~~~~ + +The ``info`` callback is used to get detailed information on this +control. This must store the values of the given +struct snd_ctl_elem_info object. For example, +for a boolean control with a single element: + +:: + + + static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + + + +The ``type`` field specifies the type of the control. There are +``BOOLEAN``, ``INTEGER``, ``ENUMERATED``, ``BYTES``, ``IEC958`` and +``INTEGER64``. The ``count`` field specifies the number of elements in +this control. For example, a stereo volume would have count = 2. The +``value`` field is a union, and the values stored are depending on the +type. The boolean and integer types are identical. + +The enumerated type is a bit different from others. You'll need to set +the string for the currently given item index. + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; + } + +The above callback can be simplified with a helper function, +:c:func:`snd_ctl_enum_info()`. The final code looks like below. +(You can pass ``ARRAY_SIZE(texts)`` instead of 4 in the third argument; +it's a matter of taste.) + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + return snd_ctl_enum_info(uinfo, 1, 4, texts); + } + + +Some common info callbacks are available for your convenience: +:c:func:`snd_ctl_boolean_mono_info()` and +:c:func:`snd_ctl_boolean_stereo_info()`. Obviously, the former +is an info callback for a mono channel boolean item, just like +:c:func:`snd_myctl_mono_info()` above, and the latter is for a +stereo channel boolean item. + +get callback +~~~~~~~~~~~~ + +This callback is used to read the current value of the control and to +return to user-space. + +For example, + +:: + + + static int snd_myctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = get_some_value(chip); + return 0; + } + + + +The ``value`` field depends on the type of control as well as on the +info callback. For example, the sb driver uses this field to store the +register offset, the bit-shift and the bit-mask. The ``private_value`` +field is set as follows: + +:: + + .private_value = reg | (shift << 16) | (mask << 24) + +and is retrieved in callbacks like + +:: + + static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } + +In the ``get`` callback, you have to fill all the elements if the +control has more than one elements, i.e. ``count > 1``. In the example +above, we filled only one element (``value.integer.value[0]``) since +it's assumed as ``count = 1``. + +put callback +~~~~~~~~~~~~ + +This callback is used to write a value from user-space. + +For example, + +:: + + + static int snd_myctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (chip->current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } + + + +As seen above, you have to return 1 if the value is changed. If the +value is not changed, return 0 instead. If any fatal error happens, +return a negative error code as usual. + +As in the ``get`` callback, when the control has more than one +elements, all elements must be evaluated in this callback, too. + +Callbacks are not atomic +~~~~~~~~~~~~~~~~~~~~~~~~ + +All these three callbacks are basically not atomic. + +Control Constructor +------------------- + +When everything is ready, finally we can create a new control. To create +a control, there are two functions to be called, +:c:func:`snd_ctl_new1()` and :c:func:`snd_ctl_add()`. + +In the simplest way, you can do like this: + +:: + + err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); + if (err < 0) + return err; + +where ``my_control`` is the struct snd_kcontrol_new object defined above, +and chip is the object pointer to be passed to kcontrol->private_data which +can be referred to in callbacks. + +:c:func:`snd_ctl_new1()` allocates a new struct snd_kcontrol instance, and +:c:func:`snd_ctl_add()` assigns the given control component to the +card. + +Change Notification +------------------- + +If you need to change and update a control in the interrupt routine, you +can call :c:func:`snd_ctl_notify()`. For example, + +:: + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); + +This function takes the card pointer, the event-mask, and the control id +pointer for the notification. The event-mask specifies the types of +notification, for example, in the above example, the change of control +values is notified. The id pointer is the pointer of struct snd_ctl_elem_id +to be notified. You can find some examples in ``es1938.c`` or ``es1968.c`` +for hardware volume interrupts. + +Metadata +-------- + +To provide information about the dB values of a mixer control, use on of +the ``DECLARE_TLV_xxx`` macros from ``<sound/tlv.h>`` to define a +variable containing this information, set the ``tlv.p`` field to point to +this variable, and include the ``SNDRV_CTL_ELEM_ACCESS_TLV_READ`` flag +in the ``access`` field; like this: + +:: + + static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); + + static struct snd_kcontrol_new my_control = { + ... + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + ... + .tlv.p = db_scale_my_control, + }; + + +The :c:func:`DECLARE_TLV_DB_SCALE()` macro defines information +about a mixer control where each step in the control's value changes the +dB value by a constant dB amount. The first parameter is the name of the +variable to be defined. The second parameter is the minimum value, in +units of 0.01 dB. The third parameter is the step size, in units of 0.01 +dB. Set the fourth parameter to 1 if the minimum value actually mutes +the control. + +The :c:func:`DECLARE_TLV_DB_LINEAR()` macro defines information +about a mixer control where the control's value affects the output +linearly. The first parameter is the name of the variable to be defined. +The second parameter is the minimum value, in units of 0.01 dB. The +third parameter is the maximum value, in units of 0.01 dB. If the +minimum value mutes the control, set the second parameter to +``TLV_DB_GAIN_MUTE``. + +API for AC97 Codec +================== + +General +------- + +The ALSA AC97 codec layer is a well-defined one, and you don't have to +write much code to control it. Only low-level control routines are +necessary. The AC97 codec API is defined in ``<sound/ac97_codec.h>``. + +Full Code Example +----------------- + +:: + + struct mychip { + .... + struct snd_ac97 *ac97; + .... + }; + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + /* read a register value here from the codec */ + return the_register_value; + } + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + { + struct mychip *chip = ac97->private_data; + .... + /* write the given register value to the codec */ + } + + static int snd_mychip_ac97(struct mychip *chip) + { + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + + +AC97 Constructor +---------------- + +To create an ac97 instance, first call :c:func:`snd_ac97_bus()` +with an ``ac97_bus_ops_t`` record with callback functions. + +:: + + struct snd_ac97_bus *bus; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + snd_ac97_bus(card, 0, &ops, NULL, &pbus); + +The bus record is shared among all belonging ac97 instances. + +And then call :c:func:`snd_ac97_mixer()` with an struct snd_ac97_template +record together with the bus pointer created above. + +:: + + struct snd_ac97_template ac97; + int err; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + snd_ac97_mixer(bus, &ac97, &chip->ac97); + +where chip->ac97 is a pointer to a newly created ``ac97_t`` +instance. In this case, the chip pointer is set as the private data, +so that the read/write callback functions can refer to this chip +instance. This instance is not necessarily stored in the chip +record. If you need to change the register values from the driver, or +need the suspend/resume of ac97 codecs, keep this pointer to pass to +the corresponding functions. + +AC97 Callbacks +-------------- + +The standard callbacks are ``read`` and ``write``. Obviously they +correspond to the functions for read and write accesses to the +hardware low-level codes. + +The ``read`` callback returns the register value specified in the +argument. + +:: + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + return the_register_value; + } + +Here, the chip can be cast from ``ac97->private_data``. + +Meanwhile, the ``write`` callback is used to set the register +value + +:: + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + + +These callbacks are non-atomic like the control API callbacks. + +There are also other callbacks: ``reset``, ``wait`` and ``init``. + +The ``reset`` callback is used to reset the codec. If the chip +requires a special kind of reset, you can define this callback. + +The ``wait`` callback is used to add some waiting time in the standard +initialization of the codec. If the chip requires the extra waiting +time, define this callback. + +The ``init`` callback is used for additional initialization of the +codec. + +Updating Registers in The Driver +-------------------------------- + +If you need to access to the codec from the driver, you can call the +following functions: :c:func:`snd_ac97_write()`, +:c:func:`snd_ac97_read()`, :c:func:`snd_ac97_update()` and +:c:func:`snd_ac97_update_bits()`. + +Both :c:func:`snd_ac97_write()` and +:c:func:`snd_ac97_update()` functions are used to set a value to +the given register (``AC97_XXX``). The difference between them is that +:c:func:`snd_ac97_update()` doesn't write a value if the given +value has been already set, while :c:func:`snd_ac97_write()` +always rewrites the value. + +:: + + snd_ac97_write(ac97, AC97_MASTER, 0x8080); + snd_ac97_update(ac97, AC97_MASTER, 0x8080); + +:c:func:`snd_ac97_read()` is used to read the value of the given +register. For example, + +:: + + value = snd_ac97_read(ac97, AC97_MASTER); + +:c:func:`snd_ac97_update_bits()` is used to update some bits in +the given register. + +:: + + snd_ac97_update_bits(ac97, reg, mask, value); + +Also, there is a function to change the sample rate (of a given register +such as ``AC97_PCM_FRONT_DAC_RATE``) when VRA or DRA is supported by the +codec: :c:func:`snd_ac97_set_rate()`. + +:: + + snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); + + +The following registers are available to set the rate: +``AC97_PCM_MIC_ADC_RATE``, ``AC97_PCM_FRONT_DAC_RATE``, +``AC97_PCM_LR_ADC_RATE``, ``AC97_SPDIF``. When ``AC97_SPDIF`` is +specified, the register is not really changed but the corresponding +IEC958 status bits will be updated. + +Clock Adjustment +---------------- + +In some chips, the clock of the codec isn't 48000 but using a PCI clock +(to save a quartz!). In this case, change the field ``bus->clock`` to +the corresponding value. For example, intel8x0 and es1968 drivers have +their own function to read from the clock. + +Proc Files +---------- + +The ALSA AC97 interface will create a proc file such as +``/proc/asound/card0/codec97#0/ac97#0-0`` and ``ac97#0-0+regs``. You +can refer to these files to see the current status and registers of +the codec. + +Multiple Codecs +--------------- + +When there are several codecs on the same card, you need to call +:c:func:`snd_ac97_mixer()` multiple times with ``ac97.num=1`` or +greater. The ``num`` field specifies the codec number. + +If you set up multiple codecs, you either need to write different +callbacks for each codec or check ``ac97->num`` in the callback +routines. + +MIDI (MPU401-UART) Interface +============================ + +General +------- + +Many soundcards have built-in MIDI (MPU401-UART) interfaces. When the +soundcard supports the standard MPU401-UART interface, most likely you +can use the ALSA MPU401-UART API. The MPU401-UART API is defined in +``<sound/mpu401.h>``. + +Some soundchips have a similar but slightly different implementation of +mpu401 stuff. For example, emu10k1 has its own mpu401 routines. + +MIDI Constructor +---------------- + +To create a rawmidi object, call :c:func:`snd_mpu401_uart_new()`. + +:: + + struct snd_rawmidi *rmidi; + snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, + irq, &rmidi); + + +The first argument is the card pointer, and the second is the index of +this component. You can create up to 8 rawmidi devices. + +The third argument is the type of the hardware, ``MPU401_HW_XXX``. If +it's not a special one, you can use ``MPU401_HW_MPU401``. + +The 4th argument is the I/O port address. Many backward-compatible +MPU401 have an I/O port such as 0x330. Or, it might be a part of its own +PCI I/O region. It depends on the chip design. + +The 5th argument is a bitflag for additional information. When the I/O +port address above is part of the PCI I/O region, the MPU401 I/O port +might have been already allocated (reserved) by the driver itself. In +such a case, pass a bit flag ``MPU401_INFO_INTEGRATED``, and the +mpu401-uart layer will allocate the I/O ports by itself. + +When the controller supports only the input or output MIDI stream, pass +the ``MPU401_INFO_INPUT`` or ``MPU401_INFO_OUTPUT`` bitflag, +respectively. Then the rawmidi instance is created as a single stream. + +``MPU401_INFO_MMIO`` bitflag is used to change the access method to MMIO +(via readb and writeb) instead of iob and outb. In this case, you have +to pass the iomapped address to :c:func:`snd_mpu401_uart_new()`. + +When ``MPU401_INFO_TX_IRQ`` is set, the output stream isn't checked in +the default interrupt handler. The driver needs to call +:c:func:`snd_mpu401_uart_interrupt_tx()` by itself to start +processing the output stream in the irq handler. + +If the MPU-401 interface shares its interrupt with the other logical +devices on the card, set ``MPU401_INFO_IRQ_HOOK`` (see +`below <MIDI Interrupt Handler_>`__). + +Usually, the port address corresponds to the command port and port + 1 +corresponds to the data port. If not, you may change the ``cport`` +field of struct snd_mpu401 manually afterward. +However, struct snd_mpu401 pointer is +not returned explicitly by :c:func:`snd_mpu401_uart_new()`. You +need to cast ``rmidi->private_data`` to struct snd_mpu401 explicitly, + +:: + + struct snd_mpu401 *mpu; + mpu = rmidi->private_data; + +and reset the ``cport`` as you like: + +:: + + mpu->cport = my_own_control_port; + +The 6th argument specifies the ISA irq number that will be allocated. If +no interrupt is to be allocated (because your code is already allocating +a shared interrupt, or because the device does not use interrupts), pass +-1 instead. For a MPU-401 device without an interrupt, a polling timer +will be used instead. + +MIDI Interrupt Handler +---------------------- + +When the interrupt is allocated in +:c:func:`snd_mpu401_uart_new()`, an exclusive ISA interrupt +handler is automatically used, hence you don't have anything else to do +than creating the mpu401 stuff. Otherwise, you have to set +``MPU401_INFO_IRQ_HOOK``, and call +:c:func:`snd_mpu401_uart_interrupt()` explicitly from your own +interrupt handler when it has determined that a UART interrupt has +occurred. + +In this case, you need to pass the private_data of the returned rawmidi +object from :c:func:`snd_mpu401_uart_new()` as the second +argument of :c:func:`snd_mpu401_uart_interrupt()`. + +:: + + snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); + + +RawMIDI Interface +================= + +Overview +-------- + +The raw MIDI interface is used for hardware MIDI ports that can be +accessed as a byte stream. It is not used for synthesizer chips that do +not directly understand MIDI. + +ALSA handles file and buffer management. All you have to do is to write +some code to move data between the buffer and the hardware. + +The rawmidi API is defined in ``<sound/rawmidi.h>``. + +RawMIDI Constructor +------------------- + +To create a rawmidi device, call the :c:func:`snd_rawmidi_new()` +function: + +:: + + struct snd_rawmidi *rmidi; + err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); + if (err < 0) + return err; + rmidi->private_data = chip; + strcpy(rmidi->name, "My MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + +The first argument is the card pointer, the second argument is the ID +string. + +The third argument is the index of this component. You can create up to +8 rawmidi devices. + +The fourth and fifth arguments are the number of output and input +substreams, respectively, of this device (a substream is the equivalent +of a MIDI port). + +Set the ``info_flags`` field to specify the capabilities of the +device. Set ``SNDRV_RAWMIDI_INFO_OUTPUT`` if there is at least one +output port, ``SNDRV_RAWMIDI_INFO_INPUT`` if there is at least one +input port, and ``SNDRV_RAWMIDI_INFO_DUPLEX`` if the device can handle +output and input at the same time. + +After the rawmidi device is created, you need to set the operators +(callbacks) for each substream. There are helper functions to set the +operators for all the substreams of a device: + +:: + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); + +The operators are usually defined like this: + +:: + + static struct snd_rawmidi_ops snd_mymidi_output_ops = { + .open = snd_mymidi_output_open, + .close = snd_mymidi_output_close, + .trigger = snd_mymidi_output_trigger, + }; + +These callbacks are explained in the `RawMIDI Callbacks`_ section. + +If there are more than one substream, you should give a unique name to +each of them: + +:: + + struct snd_rawmidi_substream *substream; + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ + +RawMIDI Callbacks +----------------- + +In all the callbacks, the private data that you've set for the rawmidi +device can be accessed as ``substream->rmidi->private_data``. + +If there is more than one port, your callbacks can determine the port +index from the struct snd_rawmidi_substream data passed to each +callback: + +:: + + struct snd_rawmidi_substream *substream; + int index = substream->number; + +RawMIDI open callback +~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_rawmidi_substream *substream); + + +This is called when a substream is opened. You can initialize the +hardware here, but you shouldn't start transmitting/receiving data yet. + +RawMIDI close callback +~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_rawmidi_substream *substream); + +Guess what. + +The ``open`` and ``close`` callbacks of a rawmidi device are +serialized with a mutex, and can sleep. + +Rawmidi trigger callback for output substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter when there is some data +in the substream buffer that must be transmitted. + +To read data from the buffer, call +:c:func:`snd_rawmidi_transmit_peek()`. It will return the number +of bytes that have been read; this will be less than the number of bytes +requested when there are no more data in the buffer. After the data have +been transmitted successfully, call +:c:func:`snd_rawmidi_transmit_ack()` to remove the data from the +substream buffer: + +:: + + unsigned char data; + while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { + if (snd_mychip_try_to_transmit(data)) + snd_rawmidi_transmit_ack(substream, 1); + else + break; /* hardware FIFO full */ + } + +If you know beforehand that the hardware will accept data, you can use +the :c:func:`snd_rawmidi_transmit()` function which reads some +data and removes them from the buffer at once: + +:: + + while (snd_mychip_transmit_possible()) { + unsigned char data; + if (snd_rawmidi_transmit(substream, &data, 1) != 1) + break; /* no more data */ + snd_mychip_transmit(data); + } + +If you know beforehand how many bytes you can accept, you can use a +buffer size greater than one with the ``snd_rawmidi_transmit*()`` functions. + +The ``trigger`` callback must not sleep. If the hardware FIFO is full +before the substream buffer has been emptied, you have to continue +transmitting data later, either in an interrupt handler, or with a +timer if the hardware doesn't have a MIDI transmit interrupt. + +The ``trigger`` callback is called with a zero ``up`` parameter when +the transmission of data should be aborted. + +RawMIDI trigger callback for input substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter to enable receiving data, +or with a zero ``up`` parameter do disable receiving data. + +The ``trigger`` callback must not sleep; the actual reading of data +from the device is usually done in an interrupt handler. + +When data reception is enabled, your interrupt handler should call +:c:func:`snd_rawmidi_receive()` for all received data: + +:: + + void snd_mychip_midi_interrupt(...) + { + while (mychip_midi_available()) { + unsigned char data; + data = mychip_midi_read(); + snd_rawmidi_receive(substream, &data, 1); + } + } + + +drain callback +~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_drain(struct snd_rawmidi_substream *substream); + + +This is only used with output substreams. This function should wait +until all data read from the substream buffer have been transmitted. +This ensures that the device can be closed and the driver unloaded +without losing data. + +This callback is optional. If you do not set ``drain`` in the struct +snd_rawmidi_ops structure, ALSA will simply wait for 50 milliseconds +instead. + +Miscellaneous Devices +===================== + +FM OPL3 +------- + +The FM OPL3 is still used in many chips (mainly for backward +compatibility). ALSA has a nice OPL3 FM control layer, too. The OPL3 API +is defined in ``<sound/opl3.h>``. + +FM registers can be directly accessed through the direct-FM API, defined +in ``<sound/asound_fm.h>``. In ALSA native mode, FM registers are +accessed through the Hardware-Dependent Device direct-FM extension API, +whereas in OSS compatible mode, FM registers can be accessed with the +OSS direct-FM compatible API in ``/dev/dmfmX`` device. + +To create the OPL3 component, you have two functions to call. The first +one is a constructor for the ``opl3_t`` instance. + +:: + + struct snd_opl3 *opl3; + snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, + integrated, &opl3); + +The first argument is the card pointer, the second one is the left port +address, and the third is the right port address. In most cases, the +right port is placed at the left port + 2. + +The fourth argument is the hardware type. + +When the left and right ports have been already allocated by the card +driver, pass non-zero to the fifth argument (``integrated``). Otherwise, +the opl3 module will allocate the specified ports by itself. + +When the accessing the hardware requires special method instead of the +standard I/O access, you can create opl3 instance separately with +:c:func:`snd_opl3_new()`. + +:: + + struct snd_opl3 *opl3; + snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); + +Then set ``command``, ``private_data`` and ``private_free`` for the +private access function, the private data and the destructor. The +``l_port`` and ``r_port`` are not necessarily set. Only the command +must be set properly. You can retrieve the data from the +``opl3->private_data`` field. + +After creating the opl3 instance via :c:func:`snd_opl3_new()`, +call :c:func:`snd_opl3_init()` to initialize the chip to the +proper state. Note that :c:func:`snd_opl3_create()` always calls +it internally. + +If the opl3 instance is created successfully, then create a hwdep device +for this opl3. + +:: + + struct snd_hwdep *opl3hwdep; + snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); + +The first argument is the ``opl3_t`` instance you created, and the +second is the index number, usually 0. + +The third argument is the index-offset for the sequencer client assigned +to the OPL3 port. When there is an MPU401-UART, give 1 for here (UART +always takes 0). + +Hardware-Dependent Devices +-------------------------- + +Some chips need user-space access for special controls or for loading +the micro code. In such a case, you can create a hwdep +(hardware-dependent) device. The hwdep API is defined in +``<sound/hwdep.h>``. You can find examples in opl3 driver or +``isa/sb/sb16_csp.c``. + +The creation of the ``hwdep`` instance is done via +:c:func:`snd_hwdep_new()`. + +:: + + struct snd_hwdep *hw; + snd_hwdep_new(card, "My HWDEP", 0, &hw); + +where the third argument is the index number. + +You can then pass any pointer value to the ``private_data``. If you +assign a private data, you should define the destructor, too. The +destructor function is set in the ``private_free`` field. + +:: + + struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); + hw->private_data = p; + hw->private_free = mydata_free; + +and the implementation of the destructor would be: + +:: + + static void mydata_free(struct snd_hwdep *hw) + { + struct mydata *p = hw->private_data; + kfree(p); + } + +The arbitrary file operations can be defined for this instance. The file +operators are defined in the ``ops`` table. For example, assume that +this chip needs an ioctl. + +:: + + hw->ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; + +And implement the callback functions as you like. + +IEC958 (S/PDIF) +--------------- + +Usually the controls for IEC958 devices are implemented via the control +interface. There is a macro to compose a name string for IEC958 +controls, :c:func:`SNDRV_CTL_NAME_IEC958()` defined in +``<include/asound.h>``. + +There are some standard controls for IEC958 status bits. These controls +use the type ``SNDRV_CTL_ELEM_TYPE_IEC958``, and the size of element is +fixed as 4 bytes array (value.iec958.status[x]). For the ``info`` +callback, you don't specify the value field for this type (the count +field must be set, though). + +“IEC958 Playback Con Mask” is used to return the bit-mask for the IEC958 +status bits of consumer mode. Similarly, “IEC958 Playback Pro Mask” +returns the bitmask for professional mode. They are read-only controls. + +Meanwhile, “IEC958 Playback Default” control is defined for getting and +setting the current default IEC958 bits. + +Due to historical reasons, both variants of the Playback Mask and the +Playback Default controls can be implemented on either a +``SNDRV_CTL_ELEM_IFACE_PCM`` or a ``SNDRV_CTL_ELEM_IFACE_MIXER`` iface. +Drivers should expose the mask and default on the same iface though. + +In addition, you can define the control switches to enable/disable or to +set the raw bit mode. The implementation will depend on the chip, but +the control should be named as “IEC958 xxx”, preferably using the +:c:func:`SNDRV_CTL_NAME_IEC958()` macro. + +You can find several cases, for example, ``pci/emu10k1``, +``pci/ice1712``, or ``pci/cmipci.c``. + +Buffer and Memory Management +============================ + +Buffer Types +------------ + +ALSA provides several different buffer allocation functions depending on +the bus and the architecture. All these have a consistent API. The +allocation of physically-contiguous pages is done via +:c:func:`snd_malloc_xxx_pages()` function, where xxx is the bus +type. + +The allocation of pages with fallback is +:c:func:`snd_malloc_xxx_pages_fallback()`. This function tries +to allocate the specified pages but if the pages are not available, it +tries to reduce the page sizes until enough space is found. + +The release the pages, call :c:func:`snd_free_xxx_pages()` +function. + +Usually, ALSA drivers try to allocate and reserve a large contiguous +physical space at the time the module is loaded for the later use. This +is called “pre-allocation”. As already written, you can call the +following function at pcm instance construction time (in the case of PCI +bus). + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &pci->dev, size, max); + +where ``size`` is the byte size to be pre-allocated and the ``max`` is +the maximum size to be changed via the ``prealloc`` proc file. The +allocator will try to get an area as large as possible within the +given size. + +The second argument (type) and the third argument (device pointer) are +dependent on the bus. For normal devices, pass the device pointer +(typically identical as ``card->dev``) to the third argument with +``SNDRV_DMA_TYPE_DEV`` type. + +For the continuous buffer unrelated to the +bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. +You can pass NULL to the device pointer in that case, which is the +default mode implying to allocate with ``GFP_KERNEL`` flag. +If you need a restricted (lower) address, set up the coherent DMA mask +bits for the device, and pass the device pointer, like the normal +device memory allocations. For this type, it's still allowed to pass +NULL to the device pointer, too, if no address restriction is needed. + +For the scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the +device pointer (see the `Non-Contiguous Buffers`_ section). + +Once the buffer is pre-allocated, you can use the allocator in the +``hw_params`` callback: + +:: + + snd_pcm_lib_malloc_pages(substream, size); + +Note that you have to pre-allocate to use this function. + +Most of drivers use, though, rather the newly introduced "managed +buffer allocation mode" instead of the manual allocation or release. +This is done by calling :c:func:`snd_pcm_set_managed_buffer_all()` +instead of :c:func:`snd_pcm_lib_preallocate_pages_for_all()`. + +:: + + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &pci->dev, size, max); + +where passed arguments are identical in both functions. +The difference in the managed mode is that PCM core will call +:c:func:`snd_pcm_lib_malloc_pages()` internally already before calling +the PCM ``hw_params`` callback, and call :c:func:`snd_pcm_lib_free_pages()` +after the PCM ``hw_free`` callback automatically. So the driver +doesn't have to call these functions explicitly in its callback any +longer. This made many driver code having NULL ``hw_params`` and +``hw_free`` entries. + +External Hardware Buffers +------------------------- + +Some chips have their own hardware buffers and the DMA transfer from the +host memory is not available. In such a case, you need to either 1) +copy/set the audio data directly to the external hardware buffer, or 2) +make an intermediate buffer and copy/set the data from it to the +external hardware buffer in interrupts (or in tasklets, preferably). + +The first case works fine if the external hardware buffer is large +enough. This method doesn't need any extra buffers and thus is more +effective. You need to define the ``copy_user`` and ``copy_kernel`` +callbacks for the data transfer, in addition to ``fill_silence`` +callback for playback. However, there is a drawback: it cannot be +mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. + +The second case allows for mmap on the buffer, although you have to +handle an interrupt or a tasklet to transfer the data from the +intermediate buffer to the hardware buffer. You can find an example in +the vxpocket driver. + +Another case is when the chip uses a PCI memory-map region for the +buffer instead of the host memory. In this case, mmap is available only +on certain architectures like the Intel one. In non-mmap mode, the data +cannot be transferred as in the normal way. Thus you need to define the +``copy_user``, ``copy_kernel`` and ``fill_silence`` callbacks as well, +as in the cases above. The examples are found in ``rme32.c`` and +``rme96.c``. + +The implementation of the ``copy_user``, ``copy_kernel`` and +``silence`` callbacks depends upon whether the hardware supports +interleaved or non-interleaved samples. The ``copy_user`` callback is +defined like below, a bit differently depending whether the direction +is playback or capture: + +:: + + static int playback_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void __user *src, unsigned long count); + static int capture_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void __user *dst, unsigned long count); + +In the case of interleaved samples, the second argument (``channel``) is +not used. The third argument (``pos``) points the current position +offset in bytes. + +The meaning of the fourth argument is different between playback and +capture. For playback, it holds the source data pointer, and for +capture, it's the destination data pointer. + +The last argument is the number of bytes to be copied. + +What you have to do in this callback is again different between playback +and capture directions. In the playback case, you copy the given amount +of data (``count``) at the specified pointer (``src``) to the specified +offset (``pos``) on the hardware buffer. When coded like memcpy-like +way, the copy would be like: + +:: + + my_memcpy_from_user(my_buffer + pos, src, count); + +For the capture direction, you copy the given amount of data (``count``) +at the specified offset (``pos``) on the hardware buffer to the +specified pointer (``dst``). + +:: + + my_memcpy_to_user(dst, my_buffer + pos, count); + +Here the functions are named as ``from_user`` and ``to_user`` because +it's the user-space buffer that is passed to these callbacks. That +is, the callback is supposed to copy from/to the user-space data +directly to/from the hardware buffer. + +Careful readers might notice that these callbacks receive the +arguments in bytes, not in frames like other callbacks. It's because +it would make coding easier like the examples above, and also it makes +easier to unify both the interleaved and non-interleaved cases, as +explained in the following. + +In the case of non-interleaved samples, the implementation will be a bit +more complicated. The callback is called for each channel, passed by +the second argument, so totally it's called for N-channels times per +transfer. + +The meaning of other arguments are almost same as the interleaved +case. The callback is supposed to copy the data from/to the given +user-space buffer, but only for the given channel. For the detailed +implementations, please check ``isa/gus/gus_pcm.c`` or +"pci/rme9652/rme9652.c" as examples. + +The above callbacks are the copy from/to the user-space buffer. There +are some cases where we want copy from/to the kernel-space buffer +instead. In such a case, ``copy_kernel`` callback is called. It'd +look like: + +:: + + static int playback_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void *src, unsigned long count); + static int capture_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void *dst, unsigned long count); + +As found easily, the only difference is that the buffer pointer is +without ``__user`` prefix; that is, a kernel-buffer pointer is passed +in the fourth argument. Correspondingly, the implementation would be +a version without the user-copy, such as: + +:: + + my_memcpy(my_buffer + pos, src, count); + +Usually for the playback, another callback ``fill_silence`` is +defined. It's implemented in a similar way as the copy callbacks +above: + +:: + + static int silence(struct snd_pcm_substream *substream, int channel, + unsigned long pos, unsigned long count); + +The meanings of arguments are the same as in the ``copy_user`` and +``copy_kernel`` callbacks, although there is no buffer pointer +argument. In the case of interleaved samples, the channel argument has +no meaning, as well as on ``copy_*`` callbacks. + +The role of ``fill_silence`` callback is to set the given amount +(``count``) of silence data at the specified offset (``pos``) on the +hardware buffer. Suppose that the data format is signed (that is, the +silent-data is 0), and the implementation using a memset-like function +would be like: + +:: + + my_memset(my_buffer + pos, 0, count); + +In the case of non-interleaved samples, again, the implementation +becomes a bit more complicated, as it's called N-times per transfer +for each channel. See, for example, ``isa/gus/gus_pcm.c``. + +Non-Contiguous Buffers +---------------------- + +If your hardware supports the page table as in emu10k1 or the buffer +descriptors as in via82xx, you can use the scatter-gather (SG) DMA. ALSA +provides an interface for handling SG-buffers. The API is provided in +``<sound/pcm.h>``. + +For creating the SG-buffer handler, call +:c:func:`snd_pcm_set_managed_buffer()` or +:c:func:`snd_pcm_set_managed_buffer_all()` with +``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI +pre-allocator. You need to pass ``&pci->dev``, where pci is +the struct pci_dev pointer of the chip as +well. + +:: + + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + &pci->dev, size, max); + +The ``struct snd_sg_buf`` instance is created as +``substream->dma_private`` in turn. You can cast the pointer like: + +:: + + struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; + +Then in :c:func:`snd_pcm_lib_malloc_pages()` call, the common SG-buffer +handler will allocate the non-contiguous kernel pages of the given size +and map them onto the virtually contiguous memory. The virtual pointer +is addressed in runtime->dma_area. The physical address +(``runtime->dma_addr``) is set to zero, because the buffer is +physically non-contiguous. The physical address table is set up in +``sgbuf->table``. You can get the physical address at a certain offset +via :c:func:`snd_pcm_sgbuf_get_addr()`. + +If you need to release the SG-buffer data explicitly, call the +standard API function :c:func:`snd_pcm_lib_free_pages()` as usual. + +Vmalloc'ed Buffers +------------------ + +It's possible to use a buffer allocated via :c:func:`vmalloc()`, for +example, for an intermediate buffer. In the recent version of kernel, +you can simply allocate it via standard +:c:func:`snd_pcm_lib_malloc_pages()` and co after setting up the +buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type. + +:: + + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); + +The NULL is passed to the device pointer argument, which indicates +that the default pages (GFP_KERNEL and GFP_HIGHMEM) will be +allocated. + +Also, note that zero is passed to both the size and the max size +arguments here. Since each vmalloc call should succeed at any time, +we don't need to pre-allocate the buffers like other continuous +pages. + +Proc Interface +============== + +ALSA provides an easy interface for procfs. The proc files are very +useful for debugging. I recommend you set up proc files if you write a +driver and want to get a running status or register dumps. The API is +found in ``<sound/info.h>``. + +To create a proc file, call :c:func:`snd_card_proc_new()`. + +:: + + struct snd_info_entry *entry; + int err = snd_card_proc_new(card, "my-file", &entry); + +where the second argument specifies the name of the proc file to be +created. The above example will create a file ``my-file`` under the +card directory, e.g. ``/proc/asound/card0/my-file``. + +Like other components, the proc entry created via +:c:func:`snd_card_proc_new()` will be registered and released +automatically in the card registration and release functions. + +When the creation is successful, the function stores a new instance in +the pointer given in the third argument. It is initialized as a text +proc file for read only. To use this proc file as a read-only text file +as it is, set the read callback with a private data via +:c:func:`snd_info_set_text_ops()`. + +:: + + snd_info_set_text_ops(entry, chip, my_proc_read); + +where the second argument (``chip``) is the private data to be used in +the callbacks. The third parameter specifies the read buffer size and +the fourth (``my_proc_read``) is the callback function, which is +defined like + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); + +In the read callback, use :c:func:`snd_iprintf()` for output +strings, which works just like normal :c:func:`printf()`. For +example, + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) + { + struct my_chip *chip = entry->private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } + +The file permissions can be changed afterwards. As default, it's set as +read only for all users. If you want to add write permission for the +user (root as default), do as follows: + +:: + + entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + +and set the write buffer size and the callback + +:: + + entry->c.text.write = my_proc_write; + +For the write callback, you can use :c:func:`snd_info_get_line()` +to get a text line, and :c:func:`snd_info_get_str()` to retrieve +a string from the line. Some examples are found in +``core/oss/mixer_oss.c``, core/oss/and ``pcm_oss.c``. + +For a raw-data proc-file, set the attributes as follows: + +:: + + static const struct snd_info_entry_ops my_file_io_ops = { + .read = my_file_io_read, + }; + + entry->content = SNDRV_INFO_CONTENT_DATA; + entry->private_data = chip; + entry->c.ops = &my_file_io_ops; + entry->size = 4096; + entry->mode = S_IFREG | S_IRUGO; + +For the raw data, ``size`` field must be set properly. This specifies +the maximum size of the proc file access. + +The read/write callbacks of raw mode are more direct than the text mode. +You need to use a low-level I/O functions such as +:c:func:`copy_from_user()` and :c:func:`copy_to_user()` to transfer the data. + +:: + + static ssize_t my_file_io_read(struct snd_info_entry *entry, + void *file_private_data, + struct file *file, + char *buf, + size_t count, + loff_t pos) + { + if (copy_to_user(buf, local_data + pos, count)) + return -EFAULT; + return count; + } + +If the size of the info entry has been set up properly, ``count`` and +``pos`` are guaranteed to fit within 0 and the given size. You don't +have to check the range in the callbacks unless any other condition is +required. + +Power Management +================ + +If the chip is supposed to work with suspend/resume functions, you need +to add power-management code to the driver. The additional code for +power-management should be ifdef-ed with ``CONFIG_PM``, or annotated +with __maybe_unused attribute; otherwise the compiler will complain +you. + +If the driver *fully* supports suspend/resume that is, the device can be +properly resumed to its state when suspend was called, you can set the +``SNDRV_PCM_INFO_RESUME`` flag in the pcm info field. Usually, this is +possible when the registers of the chip can be safely saved and restored +to RAM. If this is set, the trigger callback is called with +``SNDRV_PCM_TRIGGER_RESUME`` after the resume callback completes. + +Even if the driver doesn't support PM fully but partial suspend/resume +is still possible, it's still worthy to implement suspend/resume +callbacks. In such a case, applications would reset the status by +calling :c:func:`snd_pcm_prepare()` and restart the stream +appropriately. Hence, you can define suspend/resume callbacks below but +don't set ``SNDRV_PCM_INFO_RESUME`` info flag to the PCM. + +Note that the trigger with SUSPEND can always be called when +:c:func:`snd_pcm_suspend_all()` is called, regardless of the +``SNDRV_PCM_INFO_RESUME`` flag. The ``RESUME`` flag affects only the +behavior of :c:func:`snd_pcm_resume()`. (Thus, in theory, +``SNDRV_PCM_TRIGGER_RESUME`` isn't needed to be handled in the trigger +callback when no ``SNDRV_PCM_INFO_RESUME`` flag is set. But, it's better +to keep it for compatibility reasons.) + +In the earlier version of ALSA drivers, a common power-management layer +was provided, but it has been removed. The driver needs to define the +suspend/resume hooks according to the bus the device is connected to. In +the case of PCI drivers, the callbacks look like below: + +:: + + static int __maybe_unused snd_my_suspend(struct device *dev) + { + .... /* do things for suspend */ + return 0; + } + static int __maybe_unused snd_my_resume(struct device *dev) + { + .... /* do things for suspend */ + return 0; + } + +The scheme of the real suspend job is as follows. + +1. Retrieve the card and the chip data. + +2. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D3hot`` to change the power status. + +3. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for + each codec. + +4. Save the register values if necessary. + +5. Stop the hardware if necessary. + +A typical code would be like: + +:: + + static int __maybe_unused mychip_suspend(struct device *dev) + { + /* (1) */ + struct snd_card *card = dev_get_drvdata(dev); + struct mychip *chip = card->private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_ac97_suspend(chip->ac97); + /* (4) */ + snd_mychip_save_registers(chip); + /* (5) */ + snd_mychip_stop_hardware(chip); + return 0; + } + + +The scheme of the real resume job is as follows. + +1. Retrieve the card and the chip data. + +2. Re-initialize the chip. + +3. Restore the saved registers if necessary. + +4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. + +5. Restart the hardware (if any). + +6. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D0`` to notify the processes. + +A typical code would be like: + +:: + + static int __maybe_unused mychip_resume(struct pci_dev *pci) + { + /* (1) */ + struct snd_card *card = dev_get_drvdata(dev); + struct mychip *chip = card->private_data; + /* (2) */ + snd_mychip_reinit_chip(chip); + /* (3) */ + snd_mychip_restore_registers(chip); + /* (4) */ + snd_ac97_resume(chip->ac97); + /* (5) */ + snd_mychip_restart_chip(chip); + /* (6) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } + +Note that, at the time this callback gets called, the PCM stream has +been already suspended via its own PM ops calling +:c:func:`snd_pcm_suspend_all()` internally. + +OK, we have all callbacks now. Let's set them up. In the initialization +of the card, make sure that you can get the chip data from the card +instance, typically via ``private_data`` field, in case you created the +chip data individually. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + .... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + .... + card->private_data = chip; + .... + } + +When you created the chip data with :c:func:`snd_card_new()`, it's +anyway accessible via ``private_data`` field. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + .... + chip = card->private_data; + .... + } + +If you need a space to save the registers, allocate the buffer for it +here, too, since it would be fatal if you cannot allocate a memory in +the suspend phase. The allocated buffer should be released in the +corresponding destructor. + +And next, set suspend/resume callbacks to the pci_driver. + +:: + + static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume); + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_my_ids, + .probe = snd_my_probe, + .remove = snd_my_remove, + .driver.pm = &snd_my_pm_ops, + }; + +Module Parameters +================= + +There are standard module options for ALSA. At least, each module should +have the ``index``, ``id`` and ``enable`` options. + +If the module supports multiple cards (usually up to 8 = ``SNDRV_CARDS`` +cards), they should be arrays. The default initial values are defined +already as constants for easier programming: + +:: + + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +If the module supports only a single card, they could be single +variables, instead. ``enable`` option is not always necessary in this +case, but it would be better to have a dummy option for compatibility. + +The module parameters must be declared with the standard +``module_param()``, ``module_param_array()`` and +:c:func:`MODULE_PARM_DESC()` macros. + +The typical coding would be like below: + +:: + + #define CARD_NAME "My Chip" + + module_param_array(index, int, NULL, 0444); + MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); + module_param_array(id, charp, NULL, 0444); + MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); + module_param_array(enable, bool, NULL, 0444); + MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +Also, don't forget to define the module description and the license. +Especially, the recent modprobe requires to define the +module license as GPL, etc., otherwise the system is shown as “tainted”. + +:: + + MODULE_DESCRIPTION("Sound driver for My Chip"); + MODULE_LICENSE("GPL"); + + +Device-Managed Resources +======================== + +In the examples above, all resources are allocated and released +manually. But human beings are lazy in nature, especially developers +are lazier. So there are some ways to automate the release part; it's +the (device-)managed resources aka devres or devm family. For +example, an object allocated via :c:func:`devm_kmalloc()` will be +freed automatically at unbinding the device. + +ALSA core provides also the device-managed helper, namely, +:c:func:`snd_devm_card_new()` for creating a card object. +Call this functions instead of the normal :c:func:`snd_card_new()`, +and you can forget the explicit :c:func:`snd_card_free()` call, as +it's called automagically at error and removal paths. + +One caveat is that the call of :c:func:`snd_card_free()` would be put +at the beginning of the call chain only after you call +:c:func:`snd_card_register()`. + +Also, the ``private_free`` callback is always called at the card free, +so be careful to put the hardware clean-up procedure in +``private_free`` callback. It might be called even before you +actually set up at an earlier error path. For avoiding such an +invalid initialization, you can set ``private_free`` callback after +:c:func:`snd_card_register()` call succeeds. + +Another thing to be remarked is that you should use device-managed +helpers for each component as much as possible once when you manage +the card in that way. Mixing up with the normal and the managed +resources may screw up the release order. + + +How To Put Your Driver Into ALSA Tree +===================================== + +General +------- + +So far, you've learned how to write the driver codes. And you might have +a question now: how to put my own driver into the ALSA driver tree? Here +(finally :) the standard procedure is described briefly. + +Suppose that you create a new PCI driver for the card “xyz”. The card +module name would be snd-xyz. The new driver is usually put into the +alsa-driver tree, ``sound/pci`` directory in the case of PCI +cards. + +In the following sections, the driver code is supposed to be put into +Linux kernel tree. The two cases are covered: a driver consisting of a +single source file and one consisting of several source files. + +Driver with A Single Source File +-------------------------------- + +1. Modify sound/pci/Makefile + + Suppose you have a file xyz.c. Add the following two lines + +:: + + snd-xyz-objs := xyz.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + +2. Create the Kconfig entry + + Add the new entry of Kconfig for your xyz driver. config SND_XYZ + tristate "Foobar XYZ" depends on SND select SND_PCM help Say Y here + to include support for Foobar XYZ soundcard. To compile this driver + as a module, choose M here: the module will be called snd-xyz. the + line, select SND_PCM, specifies that the driver xyz supports PCM. In + addition to SND_PCM, the following components are supported for + select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, + SND_MPU401_UART, SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, + SND_AC97_CODEC. Add the select command for each supported + component. + + Note that some selections imply the lowlevel selections. For example, + PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC + includes PCM, and OPL3_LIB includes HWDEP. You don't need to give + the lowlevel selections again. + + For the details of Kconfig script, refer to the kbuild documentation. + +Drivers with Several Source Files +--------------------------------- + +Suppose that the driver snd-xyz have several source files. They are +located in the new subdirectory, sound/pci/xyz. + +1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile`` + as below + +:: + + obj-$(CONFIG_SND) += sound/pci/xyz/ + + +2. Under the directory ``sound/pci/xyz``, create a Makefile + +:: + + snd-xyz-objs := xyz.o abc.o def.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + +3. Create the Kconfig entry + + This procedure is as same as in the last section. + + +Useful Functions +================ + +:c:func:`snd_printk()` and friends +---------------------------------- + +.. note:: This subsection describes a few helper functions for + decorating a bit more on the standard :c:func:`printk()` & co. + However, in general, the use of such helpers is no longer recommended. + If possible, try to stick with the standard functions like + :c:func:`dev_err()` or :c:func:`pr_err()`. + +ALSA provides a verbose version of the :c:func:`printk()` function. +If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function +prints the given message together with the file name and the line of the +caller. The ``KERN_XXX`` prefix is processed as well as the original +:c:func:`printk()` does, so it's recommended to add this prefix, +e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\\n"); + +There are also :c:func:`printk()`'s for debugging. +:c:func:`snd_printd()` can be used for general debugging purposes. +If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works +just like :c:func:`snd_printk()`. If the ALSA is compiled without +the debugging flag, it's ignored. + +:c:func:`snd_printdd()` is compiled in only when +``CONFIG_SND_DEBUG_VERBOSE`` is set. + +:c:func:`snd_BUG()` +------------------- + +It shows the ``BUG?`` message and stack trace as well as +:c:func:`snd_BUG_ON()` at the point. It's useful to show that a +fatal error happens there. + +When no debug flag is set, this macro is ignored. + +:c:func:`snd_BUG_ON()` +---------------------- + +:c:func:`snd_BUG_ON()` macro is similar with +:c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or +it can be used as the condition, if (snd_BUG_ON(non_zero_is_bug)) +return -EINVAL; + +The macro takes an conditional expression to evaluate. When +``CONFIG_SND_DEBUG``, is set, if the expression is non-zero, it shows +the warning message such as ``BUG? (xxx)`` normally followed by stack +trace. In both cases it returns the evaluated value. + +Acknowledgments +=============== + +I would like to thank Phil Kerr for his help for improvement and +corrections of this document. + +Kevin Conder reformatted the original plain-text to the DocBook format. + +Giuliano Pochini corrected typos and contributed the example codes in +the hardware constraints section. diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst new file mode 100644 index 000000000..32122d687 --- /dev/null +++ b/Documentation/sound/soc/clocking.rst @@ -0,0 +1,46 @@ +============== +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/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst new file mode 100644 index 000000000..4eaa9a0c4 --- /dev/null +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -0,0 +1,113 @@ +============================================== +Creating codec to codec dai link for ALSA dapm +============================================== + +Mostly the flow of audio is always from CPU to codec so your system +will look as below: +:: + + --------- --------- + | | dai | | + CPU -------> codec + | | | | + --------- --------- + +In case your system looks as below: +:: + + --------- + | | + codec-2 + | | + --------- + | + dai-2 + | + ---------- --------- + | | dai-1 | | + CPU -------> codec-1 + | | | | + ---------- --------- + | + dai-3 + | + --------- + | | + codec-3 + | | + --------- + +Suppose codec-2 is a bluetooth chip and codec-3 is connected to +a speaker and you have a below scenario: +codec-2 will receive the audio data and the user wants to play that +audio through codec-3 without involving the CPU.This +aforementioned case is the ideal case when codec to codec +connection should be used. + +Your dai_link should appear as below in your machine +file: +:: + + /* + * this pcm stream only supports 24 bit, 2 channel and + * 48k sampling rate. + */ + static const struct snd_soc_pcm_stream dsp_codec_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }; + + { + .name = "CPU-DSP", + .stream_name = "CPU-DSP", + .cpu_dai_name = "samsung-i2s.0", + .codec_name = "codec-2, + .codec_dai_name = "codec-2-dai_name", + .platform_name = "samsung-i2s.0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dsp_codec_params, + }, + { + .name = "DSP-CODEC", + .stream_name = "DSP-CODEC", + .cpu_dai_name = "wm0010-sdi2", + .codec_name = "codec-3, + .codec_dai_name = "codec-3-dai_name", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dsp_codec_params, + }, + +Above code snippet is motivated from sound/soc/samsung/speyside.c. + +Note the "params" callback which lets the dapm know that this +dai_link is a codec to codec connection. + +In dapm core a route is created between cpu_dai playback widget +and codec_dai capture widget for playback path and vice-versa is +true for capture path. In order for this aforementioned route to get +triggered, DAPM needs to find a valid endpoint which could be either +a sink or source widget corresponding to playback and capture path +respectively. + +In order to trigger this dai_link widget, a thin codec driver for +the speaker amp can be created as demonstrated in wm8727.c file, it +sets appropriate constraints for the device even if it needs no control. + +Make sure to name your corresponding cpu and codec playback and capture +dai names ending with "Playback" and "Capture" respectively as dapm core +will link and power those dais based on the name. + +A dai_link in a "simple-audio-card" will automatically be detected as +codec to codec when all DAIs on the link belong to codec components. +The dai_link will be initialized with the subset of stream parameters +(channels, format, sample rate) supported by all DAIs on the link. Since +there is no way to provide these parameters in the device tree, this is +mostly useful for communication with simple fixed-function codecs, such +as a Bluetooth controller or cellular modem. diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst new file mode 100644 index 000000000..af973c4ca --- /dev/null +++ b/Documentation/sound/soc/codec.rst @@ -0,0 +1,190 @@ +======================= +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 +=========================== + +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, + .mute_stream = 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_rate = 1, + }; + + +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. + + +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 + + +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. +https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html + + +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.rst for details of building the description. + +Please also see the examples in other codec drivers. + + +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 */ + + +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, int direction) + { + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); + else + snd_soc_component_write(component, WM8974_DAC, mute_reg); + return 0; + } diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst new file mode 100644 index 000000000..bf8431386 --- /dev/null +++ b/Documentation/sound/soc/dai.rst @@ -0,0 +1,64 @@ +================================== +ASoC Digital Audio Interface (DAI) +================================== + +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 : +https://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, while 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:- + +I2S + MSB is transmitted on the falling edge of the first BCLK after LRC + transition. + +Left Justified + MSB is transmitted on transition of LRC. + +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 while the Tx and Rx lines are used to transmit and +receive the audio data. Bit clock usually varies depending on sample rate +while 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:- + +Mode A + MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. + +Mode B + MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst new file mode 100644 index 000000000..8e4410793 --- /dev/null +++ b/Documentation/sound/soc/dapm.rst @@ -0,0 +1,360 @@ +=================================================== +Dynamic Audio Power Management for Portable Devices +=================================================== + +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 + +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. + +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 + +Path domain + audio subsystem signal paths + + Automatically set when mixer and mux settings are changed by the user. + e.g. alsamixer, amixer. + +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. + + +DAPM Widgets +============ + +Audio DAPM widgets fall into a number of types:- + +Mixer + Mixes several analog signals into a single analog signal. +Mux + An analog switch that outputs only one of many inputs. +PGA + A programmable gain amplifier or attenuation widget. +ADC + Analog to Digital Converter +DAC + Digital to Analog Converter +Switch + An analog switch +Input + A codec input pin +Output + A codec output pin +Headphone + Headphone (and optional Jack) +Mic + Mic (and optional Jack) +Line + Line Input/Output (and optional Jack) +Speaker + Speaker +Supply + Power or clock supply widget used by other widgets. +Regulator + External regulator that supplies power to audio components. +Clock + External clock that supplies clock to audio components. +AIF IN + Audio Interface Input (with TDM slot mask). +AIF OUT + Audio Interface Output (with TDM slot mask). +Siggen + Signal Generator. +DAI IN + Digital Audio Interface Input. +DAI OUT + Digital Audio Interface Output. +DAI Link + DAI Link between two DAI structures +Pre + Special PRE widget (exec before all others) +Post + Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Scheduler + DSP internal scheduler that schedules component/pipeline processing + work. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + + +(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. + + +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), + + +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. + + +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. + +* Speaker Amp +* Microphone Bias +* 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), + + +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. + + +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(). + + +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. + + +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. + + +Endpoint Widgets +================ +An endpoint is a start or end point (widget) of an audio signal within the +machine and includes the codec. e.g. + +* Headphone Jack +* Internal Speaker +* Internal Mic +* Mic Jack +* 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. + + +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. + + +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/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst new file mode 100644 index 000000000..77f67ded5 --- /dev/null +++ b/Documentation/sound/soc/dpcm.rst @@ -0,0 +1,388 @@ +=========== +Dynamic PCM +=========== + +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 path 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(), prepare() 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. + + +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 while 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 codec 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. + + +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 convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set DAI0 to 16 bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + return 0; + } + +The other PCM operation are the same as for regular DAI links. Use as necessary. + + +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/soc/index.rst b/Documentation/sound/soc/index.rst new file mode 100644 index 000000000..e57df2dab --- /dev/null +++ b/Documentation/sound/soc/index.rst @@ -0,0 +1,20 @@ +============== +ALSA SoC Layer +============== + +The documentation is spilt into the following sections:- + +.. toctree:: + :maxdepth: 2 + + overview + codec + dai + dapm + platform + machine + pops-clicks + clocking + jack + dpcm + codec-to-codec diff --git a/Documentation/sound/soc/jack.rst b/Documentation/sound/soc/jack.rst new file mode 100644 index 000000000..644b99ecb --- /dev/null +++ b/Documentation/sound/soc/jack.rst @@ -0,0 +1,72 @@ +=================== +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/soc/machine.rst b/Documentation/sound/soc/machine.rst new file mode 100644 index 000000000..515c9444d --- /dev/null +++ b/Documentation/sound/soc/machine.rst @@ -0,0 +1,97 @@ +=================== +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/soc/overview.rst b/Documentation/sound/soc/overview.rst new file mode 100644 index 000000000..dc8370bbf --- /dev/null +++ b/Documentation/sound/soc/overview.rst @@ -0,0 +1,69 @@ +======================= +ALSA SoC Layer Overview +======================= + +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). diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst new file mode 100644 index 000000000..7036630ea --- /dev/null +++ b/Documentation/sound/soc/platform.rst @@ -0,0 +1,78 @@ +==================== +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_component_driver:- +:: + + struct snd_soc_component_driver { + const char *name; + + ... + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); + + /* pcm creation and destruction */ + int (*pcm_new)(struct snd_soc_pcm_runtime *); + void (*pcm_free)(struct snd_pcm *); + + ... + const struct snd_pcm_ops *ops; + const struct snd_compr_ops *compr_ops; + ... + }; + +Please refer to the ALSA driver documentation for details of audio DMA. +https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-driver.html + +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.rst 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/soc/pops-clicks.rst b/Documentation/sound/soc/pops-clicks.rst new file mode 100644 index 000000000..de7eb2a66 --- /dev/null +++ b/Documentation/sound/soc/pops-clicks.rst @@ -0,0 +1,55 @@ +===================== +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. |