diff options
Diffstat (limited to 'Documentation/sound')
| -rw-r--r-- | Documentation/sound/alsa-configuration.rst | 2683 | ||||
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 2330 | ||||
| -rw-r--r-- | Documentation/sound/alsa/ControlNames.txt | 107 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 324 | ||||
| -rw-r--r-- | Documentation/sound/alsa/VIA82xx-mixer.txt | 8 | ||||
| -rw-r--r-- | Documentation/sound/alsa/alsa-parameters.txt | 135 | ||||
| -rw-r--r-- | Documentation/sound/alsa/seq_oss.html | 409 | ||||
| -rw-r--r-- | Documentation/sound/cards/audigy-mixer.rst (renamed from Documentation/sound/alsa/Audigy-mixer.txt) | 297 | ||||
| -rw-r--r-- | Documentation/sound/cards/audiophile-usb.rst (renamed from Documentation/sound/alsa/Audiophile-Usb.txt) | 258 | ||||
| -rw-r--r-- | Documentation/sound/cards/bt87x.rst (renamed from Documentation/sound/alsa/Bt87x.txt) | 23 | ||||
| -rw-r--r-- | Documentation/sound/cards/cmipci.rst (renamed from Documentation/sound/alsa/CMIPCI.txt) | 62 | ||||
| -rw-r--r-- | Documentation/sound/cards/emu10k1-jack.rst (renamed from Documentation/sound/alsa/emu10k1-jack.txt) | 20 | ||||
| -rw-r--r-- | Documentation/sound/cards/hdspm.rst (renamed from Documentation/sound/alsa/hdspm.txt) | 253 | ||||
| -rw-r--r-- | Documentation/sound/cards/img-spdif-in.rst (renamed from Documentation/sound/alsa/img,spdif-in.txt) | 24 | ||||
| -rw-r--r-- | Documentation/sound/cards/index.rst | 19 | ||||
| -rw-r--r-- | Documentation/sound/cards/joystick.rst (renamed from Documentation/sound/alsa/Joystick.txt) | 71 | ||||
| -rw-r--r-- | Documentation/sound/cards/maya44.rst (renamed from Documentation/sound/alsa/README.maya44) | 137 | ||||
| -rw-r--r-- | Documentation/sound/cards/mixart.rst (renamed from Documentation/sound/alsa/MIXART.txt) | 26 | ||||
| -rw-r--r-- | Documentation/sound/cards/sb-live-mixer.rst (renamed from Documentation/sound/alsa/SB-Live-mixer.txt) | 337 | ||||
| -rw-r--r-- | Documentation/sound/cards/serial-u16550.rst (renamed from Documentation/sound/alsa/serial-u16550.txt) | 21 | ||||
| -rw-r--r-- | Documentation/sound/cards/via82xx-mixer.rst | 8 | ||||
| -rw-r--r-- | Documentation/sound/designs/channel-mapping-api.rst (renamed from Documentation/sound/alsa/Channel-Mapping-API.txt) | 77 | ||||
| -rw-r--r-- | Documentation/sound/designs/compress-offload.rst (renamed from Documentation/sound/alsa/compress_offload.txt) | 127 | ||||
| -rw-r--r-- | Documentation/sound/designs/control-names.rst | 142 | ||||
| -rw-r--r-- | Documentation/sound/designs/index.rst | 15 | ||||
| -rw-r--r-- | Documentation/sound/designs/jack-controls.rst (renamed from Documentation/sound/alsa/Jack-Controls.txt) | 13 | ||||
| -rw-r--r-- | Documentation/sound/designs/oss-emulation.rst (renamed from Documentation/sound/alsa/OSS-Emulation.txt) | 169 | ||||
| -rw-r--r-- | Documentation/sound/designs/powersave.rst (renamed from Documentation/sound/alsa/powersave.txt) | 16 | ||||
| -rw-r--r-- | Documentation/sound/designs/procfile.rst (renamed from Documentation/sound/alsa/Procfile.txt) | 106 | ||||
| -rw-r--r-- | Documentation/sound/designs/seq-oss.rst | 371 | ||||
| -rw-r--r-- | Documentation/sound/designs/timestamping.rst (renamed from Documentation/sound/alsa/timestamping.txt) | 143 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/controls.rst (renamed from Documentation/sound/alsa/HD-Audio-Controls.txt) | 33 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/dp-mst.rst (renamed from Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt) | 30 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/index.rst | 10 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/models.rst | 518 | ||||
| -rw-r--r-- | Documentation/sound/hd-audio/notes.rst (renamed from Documentation/sound/alsa/HD-Audio.txt) | 635 | ||||
| -rw-r--r-- | Documentation/sound/index.rst | 20 | ||||
| -rw-r--r-- | Documentation/sound/kernel-api/alsa-driver-api.rst | 134 | ||||
| -rw-r--r-- | Documentation/sound/kernel-api/index.rst | 8 | ||||
| -rw-r--r-- | Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 4219 | ||||
| -rw-r--r-- | Documentation/sound/soc/clocking.rst (renamed from Documentation/sound/alsa/soc/clocking.txt) | 13 | ||||
| -rw-r--r-- | Documentation/sound/soc/codec-to-codec.rst | 108 | ||||
| -rw-r--r-- | Documentation/sound/soc/codec.rst (renamed from Documentation/sound/alsa/soc/codec.txt) | 75 | ||||
| -rw-r--r-- | Documentation/sound/soc/dai.rst (renamed from Documentation/sound/alsa/soc/DAI.txt) | 28 | ||||
| -rw-r--r-- | Documentation/sound/soc/dapm.rst (renamed from Documentation/sound/alsa/soc/dapm.txt) | 249 | ||||
| -rw-r--r-- | Documentation/sound/soc/dpcm.rst (renamed from Documentation/sound/alsa/soc/DPCM.txt) | 280 | ||||
| -rw-r--r-- | Documentation/sound/soc/index.rst | 20 | ||||
| -rw-r--r-- | Documentation/sound/soc/jack.rst (renamed from Documentation/sound/alsa/soc/jack.txt) | 1 | ||||
| -rw-r--r-- | Documentation/sound/soc/machine.rst (renamed from Documentation/sound/alsa/soc/machine.txt) | 22 | ||||
| -rw-r--r-- | Documentation/sound/soc/overview.rst (renamed from Documentation/sound/alsa/soc/overview.txt) | 32 | ||||
| -rw-r--r-- | Documentation/sound/soc/platform.rst (renamed from Documentation/sound/alsa/soc/platform.txt) | 31 | ||||
| -rw-r--r-- | Documentation/sound/soc/pops-clicks.rst (renamed from Documentation/sound/alsa/soc/pops_clicks.txt) | 15 |
52 files changed, 10291 insertions, 4921 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst new file mode 100644 index 000000000000..aed6b4fb8e46 --- /dev/null +++ b/Documentation/sound/alsa-configuration.rst @@ -0,0 +1,2683 @@ +============================================================== +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 + +Common parameters for top sound card modules +-------------------------------------------- + +Each of top level sound card module takes the following options. + +index + index (slot #) of sound card; + Values: 0 through 31 or negative; + If nonnegative, assign that index number; + if negative, interpret as a bitmask of permissible indices; + the first free permitted index is assigned; + Default: -1 +id + card ID (identifier or name); + Can be up to 15 characters long; + Default: the card type; + A directory by this name is created under /proc/asound/ + containing information about the card; + This ID can be used instead of the index number in + identifying the card +enable + enable card; + Default: enabled, for PCI and ISA PnP cards + +Module snd-adlib +---------------- + +Module for AdLib FM cards. + +port + port # for OPL chip + +This module supports multiple cards. It does not support autoprobe, so +the port must be specified. For actual AdLib FM cards it will be 0x388. +Note that this card does not have PCM support and no mixer; only FM +synthesis. + +Make sure you have ``sbiload`` from the alsa-tools package available and, +after loading the module, find out the assigned ALSA sequencer port +number through ``sbiload -l``. + +Example output: +:: + + Port Client name Port name + 64:0 OPL2 FM synth OPL2 FM Port + +Load the ``std.sb`` and ``drums.sb`` patches also supplied by ``sbiload``: +:: + + sbiload -p 64:0 std.sb drums.sb + +If you use this driver to drive an OPL3, you can use ``std.o3`` and ``drums.o3`` +instead. To have the card produce sound, use ``aplaymidi`` from alsa-utils: +:: + + aplaymidi -p 64:0 foo.mid + +Module snd-ad1816a +------------------ + +Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. + +clockfreq + Clock frequency for AD1816A chip (default = 0, 33000Hz) + +This module supports multiple cards, autoprobe and PnP. + +Module snd-ad1848 +----------------- + +Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. + +port + port # for AD1848 chip +irq + IRQ # for AD1848 chip +dma1 + DMA # for AD1848 chip (0,1,3) + +This module supports multiple cards. It does not support autoprobe +thus main port must be specified!!! Other ports are optional. + +The power-management is supported. + +Module snd-ad1889 +----------------- + +Module for Analog Devices AD1889 chips. + +ac97_quirk + AC'97 workaround for strange hardware; + See the description of intel8x0 module for details. + +This module supports multiple cards. + +Module snd-ali5451 +------------------ + +Module for ALi M5451 PCI chip. + +pcm_channels + Number of hardware channels assigned for PCM +spdif + Support SPDIF I/O; + Default: disabled + +This module supports one chip and autoprobe. + +The power-management is supported. + +Module snd-als100 +----------------- + +Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. + +This module supports multiple cards, autoprobe and PnP. + +The power-management is supported. + +Module snd-als300 +----------------- + +Module for Avance Logic ALS300 and ALS300+ + +This module supports multiple cards. + +The power-management is supported. + +Module snd-als4000 +------------------ + +Module for sound cards based on Avance Logic ALS4000 PCI chip. + +joystick_port + port # for legacy joystick support; + 0 = disabled (default), 1 = auto-detect + +This module supports multiple cards, autoprobe and PnP. + +The power-management is supported. + +Module snd-asihpi +----------------- + +Module for AudioScience ASI soundcards + +enable_hpi_hwdep + enable HPI hwdep for AudioScience soundcard + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-atiixp +----------------- + +Module for ATI IXP 150/200/250/400 AC97 controllers. + +ac97_clock + AC'97 clock (default = 48000) +ac97_quirk + AC'97 workaround for strange hardware; + See `AC97 Quirk Option`_ section below. +ac97_codec + Workaround to specify which AC'97 codec instead of probing. + If this works for you file a bug with your `lspci -vn` output. + (-2 = Force probing, -1 = Default behavior, 0-2 = Use the + specified codec.) +spdif_aclink + S/PDIF transfer over AC-link (default = 1) + +This module supports one card and autoprobe. + +ATI IXP has two different methods to control SPDIF output. One is +over AC-link and another is over the "direct" SPDIF output. The +implementation depends on the motherboard, and you'll need to +choose the correct one via spdif_aclink module option. + +The power-management is supported. + +Module snd-atiixp-modem +----------------------- + +Module for ATI IXP 150/200/250 AC97 modem controllers. + +This module supports one card and autoprobe. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +The power-management is supported. + +Module snd-au8810, snd-au8820, snd-au8830 +----------------------------------------- + +Module for Aureal Vortex, Vortex2 and Advantage device. + +pcifix + Control PCI workarounds; + 0 = Disable all workarounds, + 1 = Force the PCI latency of the Aureal card to 0xff, + 2 = Force the Extend PCI#2 Internal Master for Efficient + Handling of Dummy Requests on the VIA KT133 AGP Bridge, + 3 = Force both settings, + 255 = Autodetect what is required (default) + +This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware +EQ, mpu401, gameport. A3D and wavetable support are still in development. +Development and reverse engineering work is being coordinated at +http://savannah.nongnu.org/projects/openvortex/ +SPDIF output has a copy of the AC97 codec output, unless you use the +``spdif`` pcm device, which allows raw data passthru. +The hardware EQ hardware and SPDIF is only present in the Vortex2 and +Advantage. + +Note: Some ALSA mixer applications don't handle the SPDIF sample rate +control correctly. If you have problems regarding this, try +another ALSA compliant mixer (alsamixer works). + +Module snd-azt1605 +------------------ + +Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 +chipset. + +port + port # for BASE (0x220,0x240,0x260,0x280) +wss_port + port # for WSS (0x530,0x604,0xe80,0xf40) +irq + IRQ # for WSS (7,9,10,11) +dma1 + DMA # for WSS playback (0,1,3) +dma2 + DMA # for WSS capture (0,1), -1 = disabled (default) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) +mpu_irq + IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) +fm_port + port # for OPL3 (0x388), -1 = disabled (default) + +This module supports multiple cards. It does not support autoprobe: +``port``, ``wss_port``, ``irq`` and ``dma1`` have to be specified. +The other values are optional. + +``port`` needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) +or the value stored in the card's EEPROM for cards that have an EEPROM and +their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can +be chosen freely from the options enumerated above. + +If ``dma2`` is specified and different from ``dma1``, the card will operate in +full-duplex mode. When ``dma1=3``, only ``dma2=0`` is valid and the only way to +enable capture since only channels 0 and 1 are available for capture. + +Generic settings are ``port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 +mpu_port=0x330 mpu_irq=9 fm_port=0x388``. + +Whatever IRQ and DMA channels you pick, be sure to reserve them for +legacy ISA in your BIOS. + +Module snd-azt2316 +------------------ + +Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 +chipset. + +port + port # for BASE (0x220,0x240,0x260,0x280) +wss_port + port # for WSS (0x530,0x604,0xe80,0xf40) +irq + IRQ # for WSS (7,9,10,11) +dma1 + DMA # for WSS playback (0,1,3) +dma2 + DMA # for WSS capture (0,1), -1 = disabled (default) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) +fm_port + port # for OPL3 (0x388), -1 = disabled (default) + +This module supports multiple cards. It does not support autoprobe: +``port``, ``wss_port``, ``irq`` and ``dma1`` have to be specified. +The other values are optional. + +``port`` needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) +or the value stored in the card's EEPROM for cards that have an EEPROM and +their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can +be chosen freely from the options enumerated above. + +If ``dma2`` is specified and different from ``dma1``, the card will operate in +full-duplex mode. When ``dma1=3``, only ``dma2=0`` is valid and the only way to +enable capture since only channels 0 and 1 are available for capture. + +Generic settings are ``port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 +mpu_port=0x330 mpu_irq=9 fm_port=0x388``. + +Whatever IRQ and DMA channels you pick, be sure to reserve them for +legacy ISA in your BIOS. + +Module snd-aw2 +-------------- + +Module for Audiowerk2 sound card + +This module supports multiple cards. + +Module snd-azt2320 +------------------ + +Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). + +This module supports multiple cards, PnP and autoprobe. + +The power-management is supported. + +Module snd-azt3328 +------------------ + +Module for sound cards based on Aztech AZF3328 PCI chip. + +joystick + Enable joystick (default off) + +This module supports multiple cards. + +Module snd-bt87x +---------------- + +Module for video cards based on Bt87x chips. + +digital_rate + Override the default digital rate (Hz) +load_all + Load the driver even if the card model isn't known + +This module supports multiple cards. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +Module snd-ca0106 +----------------- + +Module for Creative Audigy LS and SB Live 24bit + +This module supports multiple cards. + + +Module snd-cmi8330 +------------------ + +Module for sound cards based on C-Media CMI8330 ISA chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +wssport + port # for CMI8330 chip (WSS) +wssirq + IRQ # for CMI8330 chip (WSS) +wssdma + first DMA # for CMI8330 chip (WSS) +sbport + port # for CMI8330 chip (SB16) +sbirq + IRQ # for CMI8330 chip (SB16) +sbdma8 + 8bit DMA # for CMI8330 chip (SB16) +sbdma16 + 16bit DMA # for CMI8330 chip (SB16) +fmport + (optional) OPL3 I/O port +mpuport + (optional) MPU401 I/O port +mpuirq + (optional) MPU401 irq # + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-cmipci +----------------- + +Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. + +mpu_port + port address of MIDI interface (8338 only): + 0x300,0x310,0x320,0x330 = legacy port, + 0 = disable (default) +fm_port + port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), + 0 = disable +soft_ac3 + Software-conversion of raw SPDIF packets (model 033 only) (default = 1) +joystick_port + Joystick port address (0 = disable, 1 = auto-detect) + +This module supports autoprobe and multiple cards. + +The power-management is supported. + +Module snd-cs4231 +----------------- + +Module for sound cards based on CS4231 ISA chips. + +port + port # for CS4231 chip +mpu_port + port # for MPU-401 UART (optional), -1 = disable +irq + IRQ # for CS4231 chip +mpu_irq + IRQ # for MPU-401 UART +dma1 + first DMA # for CS4231 chip +dma2 + second DMA # for CS4231 chip + +This module supports multiple cards. This module does not support autoprobe +thus main port must be specified!!! Other ports are optional. + +The power-management is supported. + +Module snd-cs4236 +----------------- + +Module for sound cards based on CS4232/CS4232A, +CS4235/CS4236/CS4236B/CS4237B/CS4238B/CS4239 ISA chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for CS4236 chip (PnP setup - 0x534) +cport + control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) +mpu_port + port # for MPU-401 UART (PnP setup - 0x300), -1 = disable +fm_port + FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable +irq + IRQ # for CS4236 chip (5,7,9,11,12,15) +mpu_irq + IRQ # for MPU-401 UART (9,11,12,15) +dma1 + first DMA # for CS4236 chip (0,1,3) +dma2 + second DMA # for CS4236 chip (0,1,3), -1 = disable + +This module supports multiple cards. This module does not support autoprobe +(if ISA PnP is not used) thus main port and control port must be +specified!!! Other ports are optional. + +The power-management is supported. + +This module is aliased as snd-cs4232 since it provides the old +snd-cs4232 functionality, too. + +Module snd-cs4281 +----------------- + +Module for Cirrus Logic CS4281 soundchip. + +dual_codec + Secondary codec ID (0 = disable, default) + +This module supports multiple cards. + +The power-management is supported. + +Module snd-cs46xx +----------------- + +Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ +CS4624/CS4630/CS4280 PCI chips. + +external_amp + Force to enable external amplifier. +thinkpad + Force to enable Thinkpad's CLKRUN control. +mmap_valid + Support OSS mmap mode (default = 0). + +This module supports multiple cards and autoprobe. +Usually external amp and CLKRUN controls are detected automatically +from PCI sub vendor/device ids. If they don't work, give the options +above explicitly. + +The power-management is supported. + +Module snd-cs5530 +----------------- + +Module for Cyrix/NatSemi Geode 5530 chip. + +Module snd-cs5535audio +---------------------- + +Module for multifunction CS5535 companion PCI device + +The power-management is supported. + +Module snd-ctxfi +---------------- + +Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) + +* Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series +* Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series +* Creative Sound Blaster X-Fi Titanium Professional Audio +* Creative Sound Blaster X-Fi Titanium +* Creative Sound Blaster X-Fi Elite Pro +* Creative Sound Blaster X-Fi Platinum +* Creative Sound Blaster X-Fi Fatal1ty +* Creative Sound Blaster X-Fi XtremeGamer +* Creative Sound Blaster X-Fi XtremeMusic + +reference_rate + reference sample rate, 44100 or 48000 (default) +multiple + multiple to ref. sample rate, 1 or 2 (default) +subsystem + override the PCI SSID for probing; + the value consists of SSVID << 16 | SSDID. + The default is zero, which means no override. + +This module supports multiple cards. + +Module snd-darla20 +------------------ + +Module for Echoaudio Darla20 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-darla24 +------------------ + +Module for Echoaudio Darla24 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-dt019x +----------------- + +Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP +only) + +This module supports multiple cards. This module is enabled only with +ISA PnP support. + +The power-management is supported. + +Module snd-dummy +---------------- + +Module for the dummy sound card. This "card" doesn't do any output +or input, but you may use this module for any application which +requires a sound card (like RealPlayer). + +pcm_devs + Number of PCM devices assigned to each card (default = 1, up to 4) +pcm_substreams + Number of PCM substreams assigned to each PCM (default = 8, up to 128) +hrtimer + Use hrtimer (=1, default) or system timer (=0) +fake_buffer + Fake buffer allocations (default = 1) + +When multiple PCM devices are created, snd-dummy gives different +behavior to each PCM device: +* 0 = interleaved with mmap support +* 1 = non-interleaved with mmap support +* 2 = interleaved without mmap +* 3 = non-interleaved without mmap + +As default, snd-dummy drivers doesn't allocate the real buffers +but either ignores read/write or mmap a single dummy page to all +buffer pages, in order to save the resources. If your apps need +the read/ written buffer data to be consistent, pass fake_buffer=0 +option. + +The power-management is supported. + +Module snd-echo3g +----------------- + +Module for Echoaudio 3G cards (Gina3G/Layla3G) + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-emu10k1 +------------------ + +Module for EMU10K1/EMU10k2 based PCI sound cards. + +* Sound Blaster Live! +* Sound Blaster PCI 512 +* Emu APS (partially supported) +* Sound Blaster Audigy + +extin + bitmap of available external inputs for FX8010 (see bellow) +extout + bitmap of available external outputs for FX8010 (see bellow) +seq_ports + allocated sequencer ports (4 by default) +max_synth_voices + limit of voices used for wavetable (64 by default) +max_buffer_size + specifies the maximum size of wavetable/pcm buffers given in MB + unit. Default value is 128. +enable_ir + enable IR + +This module supports multiple cards and autoprobe. + +Input & Output configurations [extin/extout] +* Creative Card wo/Digital out [0x0003/0x1f03] +* Creative Card w/Digital out [0x0003/0x1f0f] +* Creative Card w/Digital CD in [0x000f/0x1f0f] +* Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] +* Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] +* Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] +* Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] +* Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] +* Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] +* Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] +* Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] +* Creative Card all ins and outs [0x3fff/0x7fff] + +The power-management is supported. + +Module snd-emu10k1x +------------------- + +Module for Creative Emu10k1X (SB Live Dell OEM version) + +This module supports multiple cards. + +Module snd-ens1370 +------------------ + +Module for Ensoniq AudioPCI ES1370 PCI sound cards. + +* SoundBlaster PCI 64 +* SoundBlaster PCI 128 + +joystick + Enable joystick (default off) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-ens1371 +------------------ + +Module for Ensoniq AudioPCI ES1371 PCI sound cards. + +* SoundBlaster PCI 64 +* SoundBlaster PCI 128 +* SoundBlaster Vibra PCI + +joystick_port + port # for joystick (0x200,0x208,0x210,0x218), 0 = disable + (default), 1 = auto-detect + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-es1688 +----------------- + +Module for ESS AudioDrive ES-1688 and ES-688 sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) +mpu_irq + IRQ # for MPU-401 port (5,7,9,10) +fm_port + port # for OPL3 (option; share the same port as default) + +with ``isapnp=0``, the following additional options are available: + +port + port # for ES-1688 chip (0x220,0x240,0x260) +irq + IRQ # for ES-1688 chip (5,7,9,10) +dma8 + DMA # for ES-1688 chip (0,1,3) + +This module supports multiple cards and autoprobe (without MPU-401 port) +and PnP with the ES968 chip. + +Module snd-es18xx +----------------- + +Module for ESS AudioDrive ES-18xx sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for ES-18xx chip (0x220,0x240,0x260) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) +fm_port + port # for FM (optional, not used) +irq + IRQ # for ES-18xx chip (5,7,9,10) +dma1 + first DMA # for ES-18xx chip (0,1,3) +dma2 + first DMA # for ES-18xx chip (0,1,3) + +This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 +port if native ISA PnP routines are not used). +When ``dma2`` is equal with ``dma1``, the driver works as half-duplex. + +The power-management is supported. + +Module snd-es1938 +----------------- + +Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-es1968 +----------------- + +Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. + +total_bufsize + total buffer size in kB (1-4096kB) +pcm_substreams_p + playback channels (1-8, default=2) +pcm_substreams_c + capture channels (1-8, default=0) +clock + clock (0 = auto-detection) +use_pm + support the power-management (0 = off, 1 = on, 2 = auto (default)) +enable_mpu + enable MPU401 (0 = off, 1 = on, 2 = auto (default)) +joystick + enable joystick (default off) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-fm801 +---------------- + +Module for ForteMedia FM801 based PCI sound cards. + +tea575x_tuner + Enable TEA575x tuner; + 1 = MediaForte 256-PCS, + 2 = MediaForte 256-PCPR, + 3 = MediaForte 64-PCR + High 16-bits are video (radio) device number + 1; + example: 0x10002 (MediaForte 256-PCPR, device 1) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-gina20 +----------------- + +Module for Echoaudio Gina20 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-gina24 +----------------- + +Module for Echoaudio Gina24 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-gusclassic +--------------------- + +Module for Gravis UltraSound Classic sound card. + +port + port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for GF1 chip (3,5,9,11,12,15) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +dma2 + DMA # for GF1 chip (1,3,5,6,7,-1=disable) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe. + +Module snd-gusextreme +--------------------- + +Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. + +port + port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) +gf1_port + port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable +irq + IRQ # for ES-1688 chip (5,7,9,10) +gf1_irq + IRQ # for GF1 chip (3,5,9,11,12,15) +mpu_irq + IRQ # for MPU-401 port (5,7,9,10) +dma8 + DMA # for ES-1688 chip (0,1,3) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe (without MPU-401 port). + +Module snd-gusmax +----------------- + +Module for Gravis UltraSound MAX sound card. + +port + port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for GF1 chip (3,5,9,11,12,15) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +dma2 + DMA # for GF1 chip (1,3,5,6,7,-1=disable) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe. + +Module snd-hda-intel +-------------------- + +Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, +PCH, SCH), ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, +RV630, RV635, RV670, RV770, VIA VT8251/VT8237A, SIS966, ULI M5461 + +[Multiple options for each card instance] + +model + force the model name +position_fix + Fix DMA pointer; + -1 = system default: choose appropriate one per controller hardware, + 0 = auto: falls back to LPIB when POSBUF doesn't work, + 1 = use LPIB, + 2 = POSBUF: use position buffer, + 3 = VIACOMBO: VIA-specific workaround for capture, + 4 = COMBO: use LPIB for playback, auto for capture stream +probe_mask + Bitmask to probe codecs (default = -1, meaning all slots); + When the bit 8 (0x100) is set, the lower 8 bits are used + as the "fixed" codec slots; i.e. the driver probes the + slots regardless what hardware reports back +probe_only + Only probing and no codec initialization (default=off); + Useful to check the initial codec status for debugging +bdl_pos_adj + Specifies the DMA IRQ timing delay in samples. + Passing -1 will make the driver to choose the appropriate + value based on the controller chip. +patch + Specifies the early "patch" files to modify the HD-audio setup + before initializing the codecs. + This option is available only when ``CONFIG_SND_HDA_PATCH_LOADER=y`` + is set. See hd-audio/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. + +If the default configuration doesn't work and one of the above +matches with your device, report it together with alsa-info.sh +output (with ``--no-upload`` option) to kernel bugzilla or alsa-devel +ML (see the section `Links and Addresses`_). + +``power_save`` and ``power_save_controller`` options are for power-saving +mode. See powersave.txt for details. + +Note 2: If you get click noises on output, try the module option +``position_fix=1`` or ``2``. ``position_fix=1`` will use the SD_LPIB +register value without FIFO size correction as the current +DMA pointer. ``position_fix=2`` will make the driver to use +the position buffer instead of reading SD_LPIB register. +(Usually SD_LPIB register is more accurate than the +position buffer.) + +``position_fix=3`` is specific to VIA devices. The position +of the capture stream is checked from both LPIB and POSBUF +values. ``position_fix=4`` is a combination mode, using LPIB +for playback and POSBUF for capture. + +NB: If you get many ``azx_get_response timeout`` messages at +loading, it's likely a problem of interrupts (e.g. ACPI irq +routing). Try to boot with options like ``pci=noacpi``. Also, you +can try ``single_cmd=1`` module option. This will switch the +communication method between HDA controller and codecs to the +single immediate commands instead of CORB/RIRB. Basically, the +single command mode is provided only for BIOS, and you won't get +unsolicited events, too. But, at least, this works independently +from the irq. Remember this is a last resort, and should be +avoided as much as possible... + +MORE NOTES ON ``azx_get_response timeout`` PROBLEMS: +On some hardware, you may need to add a proper probe_mask option +to avoid the ``azx_get_response timeout`` problem above, instead. +This occurs when the access to non-existing or non-working codec slot +(likely a modem one) causes a stall of the communication via HD-audio +bus. You can see which codec slots are probed by enabling +``CONFIG_SND_DEBUG_VERBOSE``, or simply from the file name of the codec +proc files. Then limit the slots to probe by probe_mask option. +For example, ``probe_mask=1`` means to probe only the first slot, and +``probe_mask=4`` means only the third slot. + +The power-management is supported. + +Module snd-hdsp +--------------- + +Module for RME Hammerfall DSP audio interface(s) + +This module supports multiple cards. + +Note: The firmware data can be automatically loaded via hotplug +when ``CONFIG_FW_LOADER`` is set. Otherwise, you need to load +the firmware via hdsploader utility included in alsa-tools +package. +The firmware data is found in alsa-firmware package. + +Note: snd-page-alloc module does the job which snd-hammerfall-mem +module did formerly. It will allocate the buffers in advance +when any HDSP cards are found. To make the buffer +allocation sure, load snd-page-alloc module in the early +stage of boot sequence. See `Early Buffer Allocation`_ +section. + +Module snd-hdspm +---------------- + +Module for RME HDSP MADI board. + +precise_ptr + Enable precise pointer, or disable. +line_outs_monitor + Send playback streams to analog outs by default. +enable_monitor + Enable Analog Out on Channel 63/64 by default. + +See hdspm.txt for details. + +Module snd-ice1712 +------------------ + +Module for Envy24 (ICE1712) based PCI sound cards. + +* MidiMan M Audio Delta 1010 +* MidiMan M Audio Delta 1010LT +* MidiMan M Audio Delta DiO 2496 +* MidiMan M Audio Delta 66 +* MidiMan M Audio Delta 44 +* MidiMan M Audio Delta 410 +* MidiMan M Audio Audiophile 2496 +* TerraTec EWS 88MT +* TerraTec EWS 88D +* TerraTec EWX 24/96 +* TerraTec DMX 6Fire +* TerraTec Phase 88 +* Hoontech SoundTrack DSP 24 +* Hoontech SoundTrack DSP 24 Value +* Hoontech SoundTrack DSP 24 Media 7.1 +* Event Electronics, EZ8 +* Digigram VX442 +* Lionstracs, Mediastaton +* Terrasoniq TS 88 + +model + Use the given board model, one of the following: + delta1010, dio2496, delta66, delta44, audiophile, delta410, + delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, + dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, + phase88, mediastation +omni + Omni I/O support for MidiMan M-Audio Delta44/66 +cs8427_timeout + reset timeout for the CS8427 chip (S/PDIF transceiver) in msec + resolution, default value is 500 (0.5 sec) + +This module supports multiple cards and autoprobe. +Note: The consumer part is not used with all Envy24 based cards (for +example in the MidiMan Delta 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 MIXART.txt for details. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via mixartloader utility in alsa-tools package. + +Module snd-mona +--------------- + +Module for Echoaudio Mona + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-mpu401 +----------------- + +Module for MPU-401 UART devices. + +port + port number or -1 (disable) +irq + IRQ number or -1 (disable) +pnp + PnP detection - 0 = disable, 1 = enable (default) + +This module supports multiple devices and PnP. + +Module snd-msnd-classic +----------------------- + +Module for Turtle Beach MultiSound Classic, Tahiti or Monterey +soundcards. + +io + Port # for msnd-classic card +irq + IRQ # for msnd-classic card +mem + Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000) +write_ndelay + enable write ndelay (default = 1) +calibrate_signal + calibrate signal (default = 0) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) +digital + Digital daughterboard present (default = 0) +cfg + Config port (0x250, 0x260 or 0x270) default = PnP +reset + Reset all devices +mpu_io + MPU401 I/O port +mpu_irq + MPU401 irq# +ide_io0 + IDE port #0 +ide_io1 + IDE port #1 +ide_irq + IDE irq# +joystick_io + Joystick I/O port + +The driver requires firmware files ``turtlebeach/msndinit.bin`` and +``turtlebeach/msndperm.bin`` in the proper firmware directory. + +See Documentation/sound/oss/MultiSound for important information +about this driver. Note that it has been discontinued, but the +Voyetra Turtle Beach knowledge base entry for it is still available +at +http://www.turtlebeach.com + +Module snd-msnd-pinnacle +------------------------ + +Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. + +io + Port # for pinnacle/fiji card +irq + IRQ # for pinnalce/fiji card +mem + Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000) +write_ndelay + enable write ndelay (default = 1) +calibrate_signal + calibrate signal (default = 0) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +The driver requires firmware files ``turtlebeach/pndspini.bin`` and +``turtlebeach/pndsperm.bin`` in the proper firmware directory. + +Module snd-mtpav +---------------- + +Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel +port). + +port + I/O port # for MTPAV (0x378,0x278, default=0x378) +irq + IRQ # for MTPAV (7,5, default=7) +hwports + number of supported hardware ports, default=8. + +Module supports only 1 card. This module has no enable option. + +Module snd-mts64 +---------------- + +Module for Ego Systems (ESI) Miditerminal 4140 + +This module supports multiple devices. +Requires parport (``CONFIG_PARPORT``). + +Module snd-nm256 +---------------- + +Module for NeoMagic NM256AV/ZX chips + +playback_bufsize + max playback frame size in kB (4-128kB) +capture_bufsize + max capture frame size in kB (4-128kB) +force_ac97 + 0 or 1 (disabled by default) +buffer_top + specify buffer top address +use_cache + 0 or 1 (disabled by default) +vaio_hack + alias buffer_top=0x25a800 +reset_workaround + enable AC97 RESET workaround for some laptops +reset_workaround2 + enable extended AC97 RESET workaround for some other laptops + +This module supports one chip and autoprobe. + +The power-management is supported. + +Note: on some notebooks the buffer address cannot be detected +automatically, or causes hang-up during initialization. +In such a case, specify the buffer top address explicitly via +the buffer_top option. +For example, +Sony F250: buffer_top=0x25a800 +Sony F270: buffer_top=0x272800 +The driver supports only ac97 codec. It's possible to force +to initialize/use ac97 although it's not detected. In such a +case, use ``force_ac97=1`` option - but *NO* guarantee whether it +works! + +Note: The NM256 chip can be linked internally with non-AC97 +codecs. This driver supports only the AC97 codec, and won't work +with machines with other (most likely CS423x or OPL3SAx) chips, +even though the device is detected in lspci. In such a case, try +other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP +but some doesn't have ISA PnP. You'll need to specify ``isapnp=0`` +and proper hardware parameters in the case without ISA PnP. + +Note: some laptops need a workaround for AC97 RESET. For the +known hardware like Dell Latitude LS and Sony PCG-F305, this +workaround is enabled automatically. For other laptops with a +hard freeze, you can try ``reset_workaround=1`` option. + +Note: Dell Latitude CSx laptops have another problem regarding +AC97 RESET. On these laptops, reset_workaround2 option is +turned on as default. This option is worth to try if the +previous reset_workaround option doesn't help. + +Note: This driver is really crappy. It's a porting from the +OSS driver, which is a result of black-magic reverse engineering. +The detection of codec will fail if the driver is loaded *after* +X-server as described above. You might be able to force to load +the module, but it may result in hang-up. Hence, make sure that +you load this module *before* X if you encounter this kind of +problem. + +Module snd-opl3sa2 +------------------ + +Module for Yamaha OPL3-SA2/SA3 sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + control port # for OPL3-SA chip (0x370) +sb_port + SB port # for OPL3-SA chip (0x220,0x240) +wss_port + WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) +midi_port + port # for MPU-401 UART (0x300,0x330), -1 = disable +fm_port + FM port # for OPL3-SA chip (0x388), -1 = disable +irq + IRQ # for OPL3-SA chip (5,7,9,10) +dma1 + first DMA # for Yamaha OPL3-SA chip (0,1,3) +dma2 + second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable + +This module supports multiple cards and ISA PnP. It does not support +autoprobe (if ISA PnP is not used) thus all ports must be specified!!! + +The power-management is supported. + +Module snd-opti92x-ad1848 +------------------------- + +Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. +Module works with OAK Mozart cards as well. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-opti92x-cs4231 +------------------------- + +Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) +dma2 + second DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-opti93x +------------------ + +Module for sound cards based on OPTi 82c93x chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) +dma2 + second DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-oxygen +----------------- + +Module for sound cards based on the C-Media CMI8786/8787/8788 chip: + +* Asound A-8788 +* Asus Xonar DG/DGX +* AuzenTech X-Meridian +* AuzenTech X-Meridian 2G +* Bgears b-Enspirer +* Club3D Theatron DTS +* HT-Omega Claro (plus) +* HT-Omega Claro halo (XT) +* Kuroutoshikou CMI8787-HG2PCI +* Razer Barracuda AC-1 +* Sondigo Inferno +* TempoTec HiFier Fantasia +* TempoTec HiFier Serenade + +This module supports autoprobe and multiple cards. + +Module snd-pcsp +--------------- + +Module for internal PC-Speaker. + +nopcm + Disable PC-Speaker PCM sound. Only beeps remain. +nforce_wa + enable NForce chipset workaround. Expect bad sound. + +This module supports system beeps, some kind of PCM playback and +even a few mixer controls. + +Module snd-pcxhr +---------------- + +Module for Digigram PCXHR boards + +This module supports multiple cards. + +Module snd-portman2x4 +--------------------- + +Module for Midiman Portman 2x4 parallel port MIDI interface + +This module supports multiple cards. + +Module snd-powermac (on ppc only) +--------------------------------- + +Module for PowerMac, iMac and iBook on-board soundchips + +enable_beep + enable beep using PCM (enabled as default) + +Module supports autoprobe a chip. + +Note: the driver may have problems regarding endianness. + +The power-management is supported. + +Module snd-pxa2xx-ac97 (on arm only) +------------------------------------ + +Module for AC97 driver for the Intel PXA2xx chip + +For ARM architecture only. + +The power-management is supported. + +Module snd-riptide +------------------ + +Module for Conexant Riptide chip + +joystick_port + Joystick port # (default: 0x200) +mpu_port + MPU401 port # (default: 0x330) +opl3_port + OPL3 port # (default: 0x388) + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. +You need to install the firmware file ``riptide.hex`` to the standard +firmware path (e.g. /lib/firmware). + +Module snd-rme32 +---------------- + +Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, +Prodif96 and Prodif Gold) sound cards. + +This module supports multiple cards. + +Module snd-rme96 +---------------- + +Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. + +This module supports multiple cards. + +Module snd-rme9652 +------------------ + +Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. + +precise_ptr + Enable precise pointer (doesn't work reliably). (default = 0) + +This module supports multiple cards. + +Note: snd-page-alloc module does the job which snd-hammerfall-mem +module did formerly. It will allocate the buffers in advance +when any RME9652 cards are found. To make the buffer +allocation sure, load snd-page-alloc module in the early +stage of boot sequence. See `Early Buffer Allocation`_ +section. + +Module snd-sa11xx-uda1341 (on arm only) +--------------------------------------- + +Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. + +Module supports only one card. +Module has no enable and index options. + +The power-management is supported. + +Module snd-sb8 +-------------- + +Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, SoundBlaster 2.0, +SoundBlaster Pro + +port + port # for SB DSP chip (0x220,0x240,0x260) +irq + IRQ # for SB DSP chip (5,7,9,10) +dma8 + DMA # for SB DSP chip (1,3) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-sb16 and snd-sbawe +----------------------------- + +Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), +SoundBlaster AWE 32 (PnP), SoundBlaster AWE 64 PnP + +mic_agc + Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) +csp + ASP/CSP chip support - 0 = disable (default), 1 = enable +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with isapnp=0, the following options are available: + +port + port # for SB DSP 4.x chip (0x220,0x240,0x260) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disable +awe_port + base port # for EMU8000 synthesizer (0x620,0x640,0x660) (snd-sbawe + module only) +irq + IRQ # for SB DSP 4.x chip (5,7,9,10) +dma8 + 8-bit DMA # for SB DSP 4.x chip (0,1,3) +dma16 + 16-bit DMA # for SB DSP 4.x chip (5,6,7) + +This module supports multiple cards, autoprobe and ISA PnP. + +Note: To use Vibra16X cards in 16-bit half duplex mode, you must +disable 16bit DMA with dma16 = -1 module parameter. +Also, all Sound Blaster 16 type cards can operate in 16-bit +half duplex mode through 8-bit DMA channel by disabling their +16-bit DMA channel. + +The power-management is supported. + +Module snd-sc6000 +----------------- + +Module for Gallant SC-6000 soundcard and later models: SC-6600 and +SC-7000. + +port + Port # (0x220 or 0x240) +mss_port + MSS Port # (0x530 or 0xe80) +irq + IRQ # (5,7,9,10,11) +mpu_irq + MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq +dma + DMA # (1,3,0) +joystick + Enable gameport - 0 = disable (default), 1 = enable + +This module supports multiple cards. + +This card is also known as Audio Excel DSP 16 or Zoltrix AV302. + +Module snd-sscape +----------------- + +Module for ENSONIQ SoundScape cards. + +port + Port # (PnP setup) +wss_port + WSS Port # (PnP setup) +irq + IRQ # (PnP setup) +mpu_irq + MPU-401 IRQ # (PnP setup) +dma + DMA # (PnP setup) +dma2 + 2nd DMA # (PnP setup, -1 to disable) +joystick + Enable gameport - 0 = disable (default), 1 = enable + +This module supports multiple cards. + +The driver requires the firmware loader support on kernel. + +Module snd-sun-amd7930 (on sparc only) +-------------------------------------- + +Module for AMD7930 sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-sun-cs4231 (on sparc only) +------------------------------------- + +Module for CS4231 sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-sun-dbri (on sparc only) +----------------------------------- + +Module for DBRI sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-wavefront +-------------------- + +Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + +use_cs4232_midi + Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with isapnp=0, the following options are available: + +cs4232_pcm_port + Port # for CS4232 PCM interface. +cs4232_pcm_irq + IRQ # for CS4232 PCM interface (5,7,9,11,12,15). +cs4232_mpu_port + Port # for CS4232 MPU-401 interface. +cs4232_mpu_irq + IRQ # for CS4232 MPU-401 interface (9,11,12,15). +ics2115_port + Port # for ICS2115 +ics2115_irq + IRQ # for ICS2115 +fm_port + FM OPL-3 Port # +dma1 + DMA1 # for CS4232 PCM interface. +dma2 + DMA2 # for CS4232 PCM interface. + +The below are options for wavefront_synth features: + +wf_raw + Assume that we need to boot the OS (default:no); + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. +fx_raw + Assume that the FX process needs help (default:yes); + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. +debug_default + Debug parameters for card initialization +wait_usecs + How long to wait without sleeping, usecs (default:150); + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. +sleep_interval + How long to sleep when waiting for reply (default: 100) +sleep_tries + How many times to try sleeping during a wait (default: 50) +ospath + Pathname to processed ICS2115 OS firmware (default:wavefront.os); + The path name of the ISC2115 OS firmware. In the recent + version, it's handled via firmware loader framework, so it + must be installed in the proper path, typically, + /lib/firmware. +reset_time + How long to wait for a reset to take effect (default:2) +ramcheck_time + How many seconds to wait for the RAM test (default:20) +osrun_time + How many seconds to wait for the ICS2115 OS (default:10) + +This module supports multiple cards and ISA PnP. + +Note: the firmware file ``wavefront.os`` was located in the earlier +version in /etc. Now it's loaded via firmware loader, and +must be in the proper firmware path, such as /lib/firmware. +Copy (or symlink) the file appropriately if you get an error +regarding firmware downloading after upgrading the kernel. + +Module snd-sonicvibes +--------------------- + +Module for S3 SonicVibes PCI sound cards. +* PINE Schubert 32 PCI + +reverb + Reverb Enable - 1 = enable, 0 = disable (default); + SoundCard must have onboard SRAM for this. +mge + Mic Gain Enable - 1 = enable, 0 = disable (default) + +This module supports multiple cards and autoprobe. + +Module snd-serial-u16550 +------------------------ + +Module for UART16550A serial MIDI ports. + +port + port # for UART16550A chip +irq + IRQ # for UART16550A chip, -1 = poll mode +speed + speed in bauds (9600,19200,38400,57600,115200) + 38400 = default +base + base for divisor in bauds (57600,115200,230400,460800) + 115200 = default +outs + number of MIDI ports in a serial port (1-4) + 1 = default +adaptor + Type of adaptor. + 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, + 3 = MS-124W M/B, 4 = Generic + +This module supports multiple cards. This module does not support autoprobe +thus the main port must be specified!!! Other options are optional. + +Module snd-trident +------------------ + +Module for Trident 4DWave DX/NX sound cards. +* Best Union Miss Melody 4DWave PCI +* HIS 4DWave PCI +* Warpspeed ONSpeed 4DWave PCI +* AzTech PCI 64-Q3D +* Addonics SV 750 +* CHIC True Sound 4Dwave +* Shark Predator4D-PCI +* Jaton SonicWave 4D +* SiS SI7018 PCI Audio +* Hoontech SoundTrack Digital 4DWave NX + +pcm_channels + max channels (voices) reserved for PCM +wavetable_size + max wavetable size in kB (4-?kb) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-ua101 +---------------- + +Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. + +This module supports multiple devices, autoprobe and hotplugging. + +Module snd-usb-audio +-------------------- + +Module for USB audio and USB MIDI devices. + +vid + Vendor ID for the device (optional) +pid + Product ID for the device (optional) +nrpacks + Max. number of packets per URB (default: 8) +device_setup + Device specific magic number (optional); + Influence depends on the device + Default: 0x0000 +ignore_ctl_error + Ignore any USB-controller regarding mixer interface (default: no) +autoclock + Enable auto-clock selection for UAC2 devices (default: yes) +quirk_alias + Quirk alias list, pass strings like ``0123abcd:5678beef``, which + applies the existing quirk for the device 5678:beef to a new + device 0123:abcd. + +This module supports multiple devices, autoprobe and hotplugging. + +NB: ``nrpacks`` parameter can be modified dynamically via sysfs. +Don't put the value over 20. Changing via sysfs has no sanity +check. + +NB: ``ignore_ctl_error=1`` may help when you get an error at accessing +the mixer element such as URB error -22. This happens on some +buggy USB device or the controller. + +NB: quirk_alias option is provided only for testing / development. +If you want to have a proper support, contact to upstream for +adding the matching quirk in the driver code statically. + +Module snd-usb-caiaq +-------------------- + +Module for caiaq UB audio interfaces, + +* Native Instruments RigKontrol2 +* Native Instruments Kore Controller +* Native Instruments Audio Kontrol 1 +* Native Instruments Audio 8 DJ + +This module supports multiple devices, autoprobe and hotplugging. + +Module snd-usb-usx2y +-------------------- + +Module for Tascam USB US-122, US-224 and US-428 devices. + +This module supports multiple devices, autoprobe and hotplugging. + +Note: you need to load the firmware via ``usx2yloader`` utility included +in alsa-tools and alsa-firmware packages. + +Module snd-via82xx +------------------ + +Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, 8233A, +8233C, 8235, 8237 (south) bridge. + +mpu_port + 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup + [VIA686A/686B only] +joystick + Enable joystick (default off) [VIA686A/686B only] +ac97_clock + AC'97 codec clock base (default 48000Hz) +dxs_support + support DXS channels, 0 = auto (default), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA, 5 = enable any sample rate and different + sample rates on different channels [VIA8233/C, 8235, 8237 only] +ac97_quirk + AC'97 workaround for strange hardware; + See `AC97 Quirk Option`_ section below. + +This module supports one chip and autoprobe. + +Note: on some SMP motherboards like MSI 694D the interrupts might +not be generated properly. In such a case, please try to +set the SMP (or MPS) version on BIOS to 1.1 instead of +default value 1.4. Then the interrupt number will be +assigned under 15. You might also upgrade your BIOS. + +Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) +channels as the first PCM. On these channels, up to 4 +streams can be played at the same time, and the controller +can perform sample rate conversion with separate rates for +each channel. +As default (``dxs_support = 0``), 48k fixed rate is chosen +except for the known devices since the output is often +noisy except for 48k on some mother boards due to the +bug of BIOS. +Please try once ``dxs_support=5`` and if it works on other +sample rates (e.g. 44.1kHz of mp3 playback), please let us +know the PCI subsystem vendor/device id's (output of +``lspci -nv``). +If ``dxs_support=5`` does not work, try ``dxs_support=4``; if it +doesn't work too, try dxs_support=1. (dxs_support=1 is +usually for old motherboards. The correct implemented +board should work with 4 or 5.) If it still doesn't +work and the default setting is ok, ``dxs_support=3`` is the +right choice. If the default setting doesn't work at all, +try ``dxs_support=2`` to disable the DXS channels. +In any cases, please let us know the result and the +subsystem vendor/device ids. See `Links and Addresses`_ +below. + +Note: for the MPU401 on VIA823x, use snd-mpu401 driver +additionally. The mpu_port option is for VIA686 chips only. + +The power-management is supported. + +Module snd-via82xx-modem +------------------------ + +Module for VIA82xx AC97 modem + +ac97_clock + AC'97 codec clock base (default 48000Hz) + +This module supports one card and autoprobe. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +The power-management is supported. + +Module snd-virmidi +------------------ + +Module for virtual rawmidi devices. +This module creates virtual rawmidi devices which communicate +to the corresponding ALSA sequencer ports. + +midi_devs + MIDI devices # (1-4, default=4) + +This module supports multiple cards. + +Module snd-virtuoso +------------------- + +Module for sound cards based on the Asus AV66/AV100/AV200 chips, +i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), +Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. + +This module supports autoprobe and multiple cards. + +Module snd-vx222 +---------------- + +Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. + +mic + Enable Microphone on V222 Mic (NYI) +ibl + Capture IBL size. (default = 0, minimum size) + +This module supports multiple cards. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via vxloader utility in alsa-tools package. To invoke +vxloader automatically, add the following to /etc/modprobe.d/alsa.conf + +:: + + install snd-vx222 /sbin/modprobe --first-time -i snd-vx222\ + && /usr/bin/vxloader + +(for 2.2/2.4 kernels, add ``post-install /usr/bin/vxloader`` to +/etc/modules.conf, instead.) +IBL size defines the interrupts period for PCM. The smaller size +gives smaller latency but leads to more CPU consumption, too. +The size is usually aligned to 126. As default (=0), the smallest +size is chosen. The possible IBL values can be found in +/proc/asound/cardX/vx-status proc file. + +The power-management is supported. + +Module snd-vxpocket +------------------- + +Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. + +ibl + Capture IBL size. (default = 0, minimum size) + +This module supports multiple cards. The module is compiled only when +PCMCIA is supported on kernel. + +With the older 2.6.x kernel, to activate the driver via the card +manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the +sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no +longer require a config file. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via vxloader utility in alsa-tools package. + +About capture IBL, see the description of snd-vx222 module. + +Note: snd-vxp440 driver is merged to snd-vxpocket driver since +ALSA 1.0.10. + +The power-management is supported. + +Module snd-ymfpci +----------------- + +Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). + +mpu_port + 0x300,0x330,0x332,0x334, 0 (disable) by default, + 1 (auto-detect for YMF744/754 only) +fm_port + 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default + 1 (auto-detect for YMF744/754 only) +joystick_port + 0x201,0x202,0x204,0x205, 0 (disable) by default, + 1 (auto-detect) +rear_switch + enable shared rear/line-in switch (bool) + +This module supports autoprobe and multiple chips. + +The power-management is supported. + +Module snd-pdaudiocf +-------------------- + +Module for Sound Core PDAudioCF sound card. + +The power-management is supported. + + +AC97 Quirk Option +================= + +The ac97_quirk option is used to enable/override the workaround for +specific devices on drivers for on-board AC'97 controllers like +snd-intel8x0. Some hardware have swapped output pins between Master +and Headphone, or Surround (thanks to confusion of AC'97 +specifications from version to version :-) + +The driver provides the auto-detection of known problematic devices, +but some might be unknown or wrongly detected. In such a case, pass +the proper value with this option. + +The following strings are accepted: + +default + Don't override the default setting +none + Disable the quirk +hp_only + Bind Master and Headphone controls as a single control +swap_hp + Swap headphone and master controls +swap_surround + Swap master and surround controls +ad_sharing + For AD1985, turn on OMS bit and use headphone +alc_jack + For ALC65x, turn on the jack sense mode +inv_eapd + Inverted EAPD implementation +mute_led + Bind EAPD bit for turning on/off mute LED + +For backward compatibility, the corresponding integer value -1, 0, ... +are accepted, too. + +For example, if ``Master`` volume control has no effect on your device +but only ``Headphone`` does, pass ac97_quirk=hp_only module option. + + +Configuring Non-ISAPNP Cards +============================ + +When the kernel is configured with ISA-PnP support, the modules +supporting the isapnp cards will have module options ``isapnp``. +If this option is set, *only* the ISA-PnP devices will be probed. +For probing the non ISA-PnP cards, you have to pass ``isapnp=0`` option +together with the proper i/o and irq configuration. + +When the kernel is configured without ISA-PnP support, isapnp option +will be not built in. + + +Module Autoloading Support +========================== + +The ALSA drivers can be loaded automatically on demand by defining +module aliases. The string ``snd-card-%1`` is requested for ALSA native +devices where ``%i`` is sound card number from zero to seven. + +To auto-load an ALSA driver for OSS services, define the string +``sound-slot-%i`` where ``%i`` means the slot number for OSS, which +corresponds to the card index of ALSA. Usually, define this +as the same card module. + +An example configuration for a single emu10k1 card is like below: +:: + + ----- /etc/modprobe.d/alsa.conf + alias snd-card-0 snd-emu10k1 + alias sound-slot-0 snd-emu10k1 + ----- /etc/modprobe.d/alsa.conf + +The available number of auto-loaded sound cards depends on the module +option ``cards_limit`` of snd module. As default it's set to 1. +To enable the auto-loading of multiple cards, specify the number of +sound cards in that option. + +When multiple cards are available, it'd better to specify the index +number for each card via module option, too, so that the order of +cards is kept consistent. + +An example configuration for two sound cards is like below: +:: + + ----- /etc/modprobe.d/alsa.conf + # ALSA portion + options snd cards_limit=2 + alias snd-card-0 snd-interwave + alias snd-card-1 snd-ens1371 + options snd-interwave index=0 + options snd-ens1371 index=1 + # OSS/Free portion + alias sound-slot-0 snd-interwave + alias sound-slot-1 snd-ens1371 + ----- /etc/modprobe.d/alsa.conf + +In this example, the interwave card is always loaded as the first card +(index 0) and ens1371 as the second (index 1). + +Alternative (and new) way to fixate the slot assignment is to use +``slots`` option of snd module. In the case above, specify like the +following: +:: + + options snd slots=snd-interwave,snd-ens1371 + +Then, the first slot (#0) is reserved for snd-interwave driver, and +the second (#1) for snd-ens1371. You can omit index option in each +driver if slots option is used (although you can still have them at +the same time as long as they don't conflict). + +The slots option is especially useful for avoiding the possible +hot-plugging and the resultant slot conflict. For example, in the +case above again, the first two slots are already reserved. If any +other driver (e.g. snd-usb-audio) is loaded before snd-interwave or +snd-ens1371, it will be assigned to the third or later slot. + +When a module name is given with '!', the slot will be given for any +modules but that name. For example, ``slots=!snd-pcsp`` will reserve +the first slot for any modules but snd-pcsp. + + +ALSA PCM devices to OSS devices mapping +======================================= +:: + + /dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 + /dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 + /dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 + /dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 + /dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 + /dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 + /dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 + /dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 + /dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 + +The first number from ``/dev/snd/pcmC{X}D{Y}[c|p]`` expression means +sound card number and second means device number. The ALSA devices +have either ``c`` or ``p`` suffix indicating the direction, capture and +playback, respectively. + +Please note that the device mapping above may be varied via the module +options of snd-pcm-oss module. + + +Proc interfaces (/proc/asound) +============================== + +/proc/asound/card#/pcm#[cp]/oss +------------------------------- +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 + http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt deleted file mode 100644 index fc53ccd9a629..000000000000 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ /dev/null @@ -1,2330 +0,0 @@ - - Advanced Linux Sound Architecture - Driver - ========================================== - Configuration guide - - -Kernel Configuration -==================== - -To enable ALSA support you need at least to build the kernel with -primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS, -you don't have to choose any of the OSS modules. - -Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and -PCM supports if you want to run OSS applications with ALSA. - -If you want to support the WaveTable functionality on cards such as -SB Live! then you need to enable "Sequencer support" -(CONFIG_SND_SEQUENCER). - -To make ALSA debug messages more verbose, enable the "Verbose printk" -and "Debug" options. To check for memory leaks, turn on "Debug memory" -too. "Debug detection" will add checks for the detection of cards. - -Please note that all the ALSA ISA drivers support the Linux isapnp API -(if the card supports ISA PnP). You don't need to configure the cards -using isapnptools. - - -Creating ALSA devices -===================== - -This depends on your distribution, but normally you use the /dev/MAKEDEV -script to create the necessary device nodes. On some systems you use a -script named 'snddevices'. - - -Module parameters -================= - -The user can load modules with options. If the module supports more than -one card and you have more than one card of the same type then you can -specify multiple values for the option separated by commas. - -Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. - - Module snd - ---------- - - The core ALSA module. It is used by all ALSA card drivers. - It takes the following options which have global effects. - - major - major number for sound driver - - Default: 116 - cards_limit - - limiting card index for auto-loading (1-8) - - Default: 1 - - For auto-loading more than one card, specify this - option together with snd-card-X aliases. - slots - Reserve the slot index for the given driver. - This option takes multiple strings. - See "Module Autoloading Support" section for details. - debug - Specifies the debug message level - (0 = disable debug prints, 1 = normal debug messages, - 2 = verbose debug messages) - This option appears only when CONFIG_SND_DEBUG=y. - This option can be dynamically changed via sysfs - /sys/modules/snd/parameters/debug file. - - Module snd-pcm-oss - ------------------ - - The PCM OSS emulation module. - This module takes options which change the mapping of devices. - - dsp_map - PCM device number maps assigned to the 1st OSS device. - - Default: 0 - adsp_map - PCM device number maps assigned to the 2st OSS device. - - Default: 1 - nonblock_open - - Don't block opening busy PCM devices. Default: 1 - - For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of - the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped - to PCM #0 of the card #0. - For changing the second or later card, specify the option with - commas, such like "dsp_map=0,1". - - nonblock_open option is used to change the behavior of the PCM - regarding opening the device. When this option is non-zero, - opening a busy OSS PCM device won't be blocked but return - immediately with EAGAIN (just like O_NONBLOCK flag). - - Module snd-rawmidi - ------------------ - - This module takes options which change the mapping of devices. - similar to those of the snd-pcm-oss module. - - midi_map - MIDI device number maps assigned to the 1st OSS device. - - Default: 0 - amidi_map - MIDI device number maps assigned to the 2st OSS device. - - Default: 1 - - Common parameters for top sound card modules - -------------------------------------------- - - Each of top level sound card module takes the following options. - - index - index (slot #) of sound card - - Values: 0 through 31 or negative - - If nonnegative, assign that index number - - if negative, interpret as a bitmask of permissible - indices; the first free permitted index is assigned - - Default: -1 - id - card ID (identifier or name) - - Can be up to 15 characters long - - Default: the card type - - A directory by this name is created under /proc/asound/ - containing information about the card - - This ID can be used instead of the index number in - identifying the card - enable - enable card - - Default: enabled, for PCI and ISA PnP cards - - Module snd-adlib - ---------------- - - Module for AdLib FM cards. - - port - port # for OPL chip - - This module supports multiple cards. It does not support autoprobe, so - the port must be specified. For actual AdLib FM cards it will be 0x388. - Note that this card does not have PCM support and no mixer; only FM - synthesis. - - Make sure you have "sbiload" from the alsa-tools package available and, - after loading the module, find out the assigned ALSA sequencer port - number through "sbiload -l". Example output: - - Port Client name Port name - 64:0 OPL2 FM synth OPL2 FM Port - - Load the std.sb and drums.sb patches also supplied by sbiload: - - sbiload -p 64:0 std.sb drums.sb - - If you use this driver to drive an OPL3, you can use std.o3 and drums.o3 - instead. To have the card produce sound, use aplaymidi from alsa-utils: - - aplaymidi -p 64:0 foo.mid - - Module snd-ad1816a - ------------------ - - Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. - - clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) - - This module supports multiple cards, autoprobe and PnP. - - Module snd-ad1848 - ----------------- - - Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. - - port - port # for AD1848 chip - irq - IRQ # for AD1848 chip - dma1 - DMA # for AD1848 chip (0,1,3) - - This module supports multiple cards. It does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-ad1889 - ----------------- - - Module for Analog Devices AD1889 chips. - - ac97_quirk - AC'97 workaround for strange hardware - See the description of intel8x0 module for details. - - This module supports multiple cards. - - Module snd-ali5451 - ------------------ - - Module for ALi M5451 PCI chip. - - pcm_channels - Number of hardware channels assigned for PCM - spdif - Support SPDIF I/O - - Default: disabled - - This module supports one chip and autoprobe. - - The power-management is supported. - - Module snd-als100 - ----------------- - - Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-als300 - ----------------- - - Module for Avance Logic ALS300 and ALS300+ - - This module supports multiple cards. - - The power-management is supported. - - Module snd-als4000 - ------------------ - - Module for sound cards based on Avance Logic ALS4000 PCI chip. - - joystick_port - port # for legacy joystick support. - 0 = disabled (default), 1 = auto-detect - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-asihpi - ----------------- - - Module for AudioScience ASI soundcards - - enable_hpi_hwdep - enable HPI hwdep for AudioScience soundcard - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-atiixp - ----------------- - - Module for ATI IXP 150/200/250/400 AC97 controllers. - - ac97_clock - AC'97 clock (default = 48000) - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - ac97_codec - Workaround to specify which AC'97 codec - instead of probing. If this works for you - file a bug with your `lspci -vn` output. - -2 -- Force probing. - -1 -- Default behavior. - 0-2 -- Use the specified codec. - spdif_aclink - S/PDIF transfer over AC-link (default = 1) - - This module supports one card and autoprobe. - - ATI IXP has two different methods to control SPDIF output. One is - over AC-link and another is over the "direct" SPDIF output. The - implementation depends on the motherboard, and you'll need to - choose the correct one via spdif_aclink module option. - - The power-management is supported. - - Module snd-atiixp-modem - ----------------------- - - Module for ATI IXP 150/200/250 AC97 modem controllers. - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-au8810, snd-au8820, snd-au8830 - ----------------------------------------- - - Module for Aureal Vortex, Vortex2 and Advantage device. - - pcifix - Control PCI workarounds - 0 = Disable all workarounds - 1 = Force the PCI latency of the Aureal card to 0xff - 2 = Force the Extend PCI#2 Internal Master for Efficient - Handling of Dummy Requests on the VIA KT133 AGP Bridge - 3 = Force both settings - 255 = Autodetect what is required (default) - - This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware - EQ, mpu401, gameport. A3D and wavetable support are still in development. - Development and reverse engineering work is being coordinated at - http://savannah.nongnu.org/projects/openvortex/ - SPDIF output has a copy of the AC97 codec output, unless you use the - "spdif" pcm device, which allows raw data passthru. - The hardware EQ hardware and SPDIF is only present in the Vortex2 and - Advantage. - - Note: Some ALSA mixer applications don't handle the SPDIF sample rate - control correctly. If you have problems regarding this, try - another ALSA compliant mixer (alsamixer works). - - Module snd-azt1605 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-azt2316 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-aw2 - -------------- - - Module for Audiowerk2 sound card - - This module supports multiple cards. - - Module snd-azt2320 - ------------------ - - Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). - - This module supports multiple cards, PnP and autoprobe. - - The power-management is supported. - - Module snd-azt3328 - ------------------ - - Module for sound cards based on Aztech AZF3328 PCI chip. - - joystick - Enable joystick (default off) - - This module supports multiple cards. - - Module snd-bt87x - ---------------- - - Module for video cards based on Bt87x chips. - - digital_rate - Override the default digital rate (Hz) - load_all - Load the driver even if the card model isn't known - - This module supports multiple cards. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - Module snd-ca0106 - ----------------- - - Module for Creative Audigy LS and SB Live 24bit - - This module supports multiple cards. - - - Module snd-cmi8330 - ------------------ - - Module for sound cards based on C-Media CMI8330 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - wssport - port # for CMI8330 chip (WSS) - wssirq - IRQ # for CMI8330 chip (WSS) - wssdma - first DMA # for CMI8330 chip (WSS) - sbport - port # for CMI8330 chip (SB16) - sbirq - IRQ # for CMI8330 chip (SB16) - sbdma8 - 8bit DMA # for CMI8330 chip (SB16) - sbdma16 - 16bit DMA # for CMI8330 chip (SB16) - fmport - (optional) OPL3 I/O port - mpuport - (optional) MPU401 I/O port - mpuirq - (optional) MPU401 irq # - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-cmipci - ----------------- - - Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. - - mpu_port - port address of MIDI interface (8338 only): - 0x300,0x310,0x320,0x330 = legacy port, - 0 = disable (default) - fm_port - port address of OPL-3 FM synthesizer (8x38 only): - 0x388 = legacy port, - 1 = integrated PCI port (default on 8738), - 0 = disable - soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) - (default = 1) - joystick_port - Joystick port address (0 = disable, 1 = auto-detect) - - This module supports autoprobe and multiple cards. - - The power-management is supported. - - Module snd-cs4231 - ----------------- - - Module for sound cards based on CS4231 ISA chips. - - port - port # for CS4231 chip - mpu_port - port # for MPU-401 UART (optional), -1 = disable - irq - IRQ # for CS4231 chip - mpu_irq - IRQ # for MPU-401 UART - dma1 - first DMA # for CS4231 chip - dma2 - second DMA # for CS4231 chip - - This module supports multiple cards. This module does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-cs4236 - ----------------- - - Module for sound cards based on CS4232/CS4232A, - CS4235/CS4236/CS4236B/CS4237B/ - CS4238B/CS4239 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for CS4236 chip (PnP setup - 0x534) - cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) - mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable - fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable - irq - IRQ # for CS4236 chip (5,7,9,11,12,15) - mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) - dma1 - first DMA # for CS4236 chip (0,1,3) - dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable - - This module supports multiple cards. This module does not support autoprobe - (if ISA PnP is not used) thus main port and control port must be - specified!!! Other ports are optional. - - The power-management is supported. - - This module is aliased as snd-cs4232 since it provides the old - snd-cs4232 functionality, too. - - Module snd-cs4281 - ----------------- - - Module for Cirrus Logic CS4281 soundchip. - - dual_codec - Secondary codec ID (0 = disable, default) - - This module supports multiple cards. - - The power-management is supported. - - Module snd-cs46xx - ----------------- - - Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ - CS4624/CS4630/CS4280 PCI chips. - - external_amp - Force to enable external amplifier. - thinkpad - Force to enable Thinkpad's CLKRUN control. - mmap_valid - Support OSS mmap mode (default = 0). - - This module supports multiple cards and autoprobe. - Usually external amp and CLKRUN controls are detected automatically - from PCI sub vendor/device ids. If they don't work, give the options - above explicitly. - - The power-management is supported. - - Module snd-cs5530 - _________________ - - Module for Cyrix/NatSemi Geode 5530 chip. - - Module snd-cs5535audio - ---------------------- - - Module for multifunction CS5535 companion PCI device - - The power-management is supported. - - Module snd-ctxfi - ---------------- - - Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) - * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series - * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series - * Creative Sound Blaster X-Fi Titanium Professional Audio - * Creative Sound Blaster X-Fi Titanium - * Creative Sound Blaster X-Fi Elite Pro - * Creative Sound Blaster X-Fi Platinum - * Creative Sound Blaster X-Fi Fatal1ty - * Creative Sound Blaster X-Fi XtremeGamer - * Creative Sound Blaster X-Fi XtremeMusic - - reference_rate - reference sample rate, 44100 or 48000 (default) - multiple - multiple to ref. sample rate, 1 or 2 (default) - subsystem - override the PCI SSID for probing; the value - consists of SSVID << 16 | SSDID. The default is - zero, which means no override. - - This module supports multiple cards. - - Module snd-darla20 - ------------------ - - Module for Echoaudio Darla20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-darla24 - ------------------ - - Module for Echoaudio Darla24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-dt019x - ----------------- - - Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP - only) - - This module supports multiple cards. This module is enabled only with - ISA PnP support. - - The power-management is supported. - - Module snd-dummy - ---------------- - - Module for the dummy sound card. This "card" doesn't do any output - or input, but you may use this module for any application which - requires a sound card (like RealPlayer). - - pcm_devs - Number of PCM devices assigned to each card - (default = 1, up to 4) - pcm_substreams - Number of PCM substreams assigned to each PCM - (default = 8, up to 128) - hrtimer - Use hrtimer (=1, default) or system timer (=0) - fake_buffer - Fake buffer allocations (default = 1) - - When multiple PCM devices are created, snd-dummy gives different - behavior to each PCM device: - 0 = interleaved with mmap support - 1 = non-interleaved with mmap support - 2 = interleaved without mmap - 3 = non-interleaved without mmap - - As default, snd-dummy drivers doesn't allocate the real buffers - but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the resources. If your apps need - the read/ written buffer data to be consistent, pass fake_buffer=0 - option. - - The power-management is supported. - - Module snd-echo3g - ----------------- - - Module for Echoaudio 3G cards (Gina3G/Layla3G) - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-emu10k1 - ------------------ - - Module for EMU10K1/EMU10k2 based PCI sound cards. - * Sound Blaster Live! - * Sound Blaster PCI 512 - * Emu APS (partially supported) - * Sound Blaster Audigy - - extin - bitmap of available external inputs for FX8010 (see bellow) - extout - bitmap of available external outputs for FX8010 (see bellow) - seq_ports - allocated sequencer ports (4 by default) - max_synth_voices - limit of voices used for wavetable (64 by default) - max_buffer_size - specifies the maximum size of wavetable/pcm buffers - given in MB unit. Default value is 128. - enable_ir - enable IR - - This module supports multiple cards and autoprobe. - - Input & Output configurations [extin/extout] - * Creative Card wo/Digital out [0x0003/0x1f03] - * Creative Card w/Digital out [0x0003/0x1f0f] - * Creative Card w/Digital CD in [0x000f/0x1f0f] - * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] - * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] - * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] - * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] - * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] - * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] - * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] - * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] - * Creative Card all ins and outs [0x3fff/0x7fff] - - The power-management is supported. - - Module snd-emu10k1x - ------------------- - - Module for Creative Emu10k1X (SB Live Dell OEM version) - - This module supports multiple cards. - - Module snd-ens1370 - ------------------ - - Module for Ensoniq AudioPCI ES1370 PCI sound cards. - * SoundBlaster PCI 64 - * SoundBlaster PCI 128 - - joystick - Enable joystick (default off) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-ens1371 - ------------------ - - Module for Ensoniq AudioPCI ES1371 PCI sound cards. - * SoundBlaster PCI 64 - * SoundBlaster PCI 128 - * SoundBlaster Vibra PCI - - joystick_port - port # for joystick (0x200,0x208,0x210,0x218), - 0 = disable (default), 1 = auto-detect - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-es1688 - ----------------- - - Module for ESS AudioDrive ES-1688 and ES-688 sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - mpu_irq - IRQ # for MPU-401 port (5,7,9,10) - fm_port - port # for OPL3 (option; share the same port as default) - - with isapnp=0, the following additional options are available: - port - port # for ES-1688 chip (0x220,0x240,0x260) - irq - IRQ # for ES-1688 chip (5,7,9,10) - dma8 - DMA # for ES-1688 chip (0,1,3) - - This module supports multiple cards and autoprobe (without MPU-401 port) - and PnP with the ES968 chip. - - Module snd-es18xx - ----------------- - - Module for ESS AudioDrive ES-18xx sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for ES-18xx chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - fm_port - port # for FM (optional, not used) - irq - IRQ # for ES-18xx chip (5,7,9,10) - dma1 - first DMA # for ES-18xx chip (0,1,3) - dma2 - first DMA # for ES-18xx chip (0,1,3) - - This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 - port if native ISA PnP routines are not used). - When dma2 is equal with dma1, the driver works as half-duplex. - - The power-management is supported. - - Module snd-es1938 - ----------------- - - Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-es1968 - ----------------- - - Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. - - total_bufsize - total buffer size in kB (1-4096kB) - pcm_substreams_p - playback channels (1-8, default=2) - pcm_substreams_c - capture channels (1-8, default=0) - clock - clock (0 = auto-detection) - use_pm - support the power-management (0 = off, 1 = on, - 2 = auto (default)) - enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default)) - joystick - enable joystick (default off) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-fm801 - ---------------- - - Module for ForteMedia FM801 based PCI sound cards. - - tea575x_tuner - Enable TEA575x tuner - - 1 = MediaForte 256-PCS - - 2 = MediaForte 256-PCPR - - 3 = MediaForte 64-PCR - - High 16-bits are video (radio) device number + 1 - - example: 0x10002 (MediaForte 256-PCPR, device 1) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-gina20 - ----------------- - - Module for Echoaudio Gina20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-gina24 - ----------------- - - Module for Echoaudio Gina24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-gusclassic - --------------------- - - Module for Gravis UltraSound Classic sound card. - - port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for GF1 chip (3,5,9,11,12,15) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe. - - Module snd-gusextreme - --------------------- - - Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. - - port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) - gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable - irq - IRQ # for ES-1688 chip (5,7,9,10) - gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15) - mpu_irq - IRQ # for MPU-401 port (5,7,9,10) - dma8 - DMA # for ES-1688 chip (0,1,3) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe (without MPU-401 port). - - Module snd-gusmax - ----------------- - - Module for Gravis UltraSound MAX sound card. - - port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for GF1 chip (3,5,9,11,12,15) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe. - - Module snd-hda-intel - -------------------- - - Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, - PCH, SCH), - ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, - RV630, RV635, RV670, RV770, - VIA VT8251/VT8237A, - SIS966, ULI M5461 - - [Multiple options for each card instance] - model - force the model name - position_fix - Fix DMA pointer - -1 = system default: choose appropriate one per controller - hardware - 0 = auto: falls back to LPIB when POSBUF doesn't work - 1 = use LPIB - 2 = POSBUF: use position buffer - 3 = VIACOMBO: VIA-specific workaround for capture - 4 = COMBO: use LPIB for playback, auto for capture stream - probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) - When the bit 8 (0x100) is set, the lower 8 bits are used - as the "fixed" codec slots; i.e. the driver probes the - slots regardless what hardware reports back - probe_only - Only probing and no codec initialization (default=off); - Useful to check the initial codec status for debugging - bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. - Passing -1 will make the driver to choose the appropriate - value based on the controller chip. - patch - Specifies the early "patch" files to modify the HD-audio - setup before initializing the codecs. This option is - available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. - See HD-Audio.txt for details. - beep_mode - Selects the beep registration mode (0=off, 1=on); default - value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig. - - [Single (global) options] - single_cmd - Use single immediate commands to communicate with - codecs (for debugging only) - enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) - power_save - Automatic power-saving timeout (in second, 0 = - disable) - power_save_controller - Reset HD-audio controller in power-saving mode - (default = on) - align_buffer_size - Force rounding of buffer/period sizes to multiples - of 128 bytes. This is more efficient in terms of memory - access but isn't required by the HDA spec and prevents - users from specifying exact period/buffer sizes. - (default = on) - snoop - Enable/disable snooping (default = on) - - This module supports multiple cards and autoprobe. - - See Documentation/sound/alsa/HD-Audio.txt for more details about - HD-audio driver. - - Each codec may have a model table for different configurations. - If your machine isn't listed there, the default (usually minimal) - configuration is set up. You can pass "model=<name>" option to - specify a certain model in such a case. There are different - models depending on the codec chip. The list of available models - is found in HD-Audio-Models.txt - - The model name "generic" is treated as a special case. When this - model is given, the driver uses the generic codec parser without - "codec-patch". It's sometimes good for testing and debugging. - - If the default configuration doesn't work and one of the above - matches with your device, report it together with alsa-info.sh - output (with --no-upload option) to kernel bugzilla or alsa-devel - ML (see the section "Links and Addresses"). - - power_save and power_save_controller options are for power-saving - mode. See powersave.txt for details. - - Note 2: If you get click noises on output, try the module option - position_fix=1 or 2. position_fix=1 will use the SD_LPIB - register value without FIFO size correction as the current - DMA pointer. position_fix=2 will make the driver to use - the position buffer instead of reading SD_LPIB register. - (Usually SD_LPIB register is more accurate than the - position buffer.) - - position_fix=3 is specific to VIA devices. The position - of the capture stream is checked from both LPIB and POSBUF - values. position_fix=4 is a combination mode, using LPIB - for playback and POSBUF for capture. - - NB: If you get many "azx_get_response timeout" messages at - loading, it's likely a problem of interrupts (e.g. ACPI irq - routing). Try to boot with options like "pci=noacpi". Also, you - can try "single_cmd=1" module option. This will switch the - communication method between HDA controller and codecs to the - single immediate commands instead of CORB/RIRB. Basically, the - single command mode is provided only for BIOS, and you won't get - unsolicited events, too. But, at least, this works independently - from the irq. Remember this is a last resort, and should be - avoided as much as possible... - - MORE NOTES ON "azx_get_response timeout" PROBLEMS: - On some hardware, you may need to add a proper probe_mask option - to avoid the "azx_get_response timeout" problem above, instead. - This occurs when the access to non-existing or non-working codec slot - (likely a modem one) causes a stall of the communication via HD-audio - bus. You can see which codec slots are probed by enabling - CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec - proc files. Then limit the slots to probe by probe_mask option. - For example, probe_mask=1 means to probe only the first slot, and - probe_mask=4 means only the third slot. - - The power-management is supported. - - Module snd-hdsp - --------------- - - Module for RME Hammerfall DSP audio interface(s) - - This module supports multiple cards. - - Note: The firmware data can be automatically loaded via hotplug - when CONFIG_FW_LOADER is set. Otherwise, you need to load - the firmware via hdsploader utility included in alsa-tools - package. - The firmware data is found in alsa-firmware package. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any HDSP cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-hdspm - ---------------- - - Module for RME HDSP MADI board. - - precise_ptr - Enable precise pointer, or disable. - line_outs_monitor - Send playback streams to analog outs by default. - enable_monitor - Enable Analog Out on Channel 63/64 by default. - - See hdspm.txt for details. - - Module snd-ice1712 - ------------------ - - Module for Envy24 (ICE1712) based PCI sound cards. - * MidiMan M Audio Delta 1010 - * MidiMan M Audio Delta 1010LT - * MidiMan M Audio Delta DiO 2496 - * MidiMan M Audio Delta 66 - * MidiMan M Audio Delta 44 - * MidiMan M Audio Delta 410 - * MidiMan M Audio Audiophile 2496 - * TerraTec EWS 88MT - * TerraTec EWS 88D - * TerraTec EWX 24/96 - * TerraTec DMX 6Fire - * TerraTec Phase 88 - * Hoontech SoundTrack DSP 24 - * Hoontech SoundTrack DSP 24 Value - * Hoontech SoundTrack DSP 24 Media 7.1 - * Event Electronics, EZ8 - * Digigram VX442 - * Lionstracs, Mediastaton - * Terrasoniq TS 88 - - model - Use the given board model, one of the following: - delta1010, dio2496, delta66, delta44, audiophile, delta410, - delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, - dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, - phase88, mediastation - omni - Omni I/O support for MidiMan M-Audio Delta44/66 - cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver) - in msec resolution, default value is 500 (0.5 sec) - - This module supports multiple cards and autoprobe. Note: The consumer part - is not used with all Envy24 based cards (for example in the MidiMan Delta - serie). - - Note: The supported board is detected by reading EEPROM or PCI - SSID (if EEPROM isn't available). You can override the - model by passing "model" module option in case that the - driver isn't configured properly or you want to try another - type for testing. - - Module snd-ice1724 - ------------------ - - Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. - * MidiMan M Audio Revolution 5.1 - * MidiMan M Audio Revolution 7.1 - * MidiMan M Audio Audiophile 192 - * AMP Ltd AUDIO2000 - * TerraTec Aureon 5.1 Sky - * TerraTec Aureon 7.1 Space - * TerraTec Aureon 7.1 Universe - * TerraTec Phase 22 - * TerraTec Phase 28 - * AudioTrak Prodigy 7.1 - * AudioTrak Prodigy 7.1 LT - * AudioTrak Prodigy 7.1 XT - * AudioTrak Prodigy 7.1 HIFI - * AudioTrak Prodigy 7.1 HD2 - * AudioTrak Prodigy 192 - * Pontis MS300 - * Albatron K8X800 Pro II - * Chaintech ZNF3-150 - * Chaintech ZNF3-250 - * Chaintech 9CJS - * Chaintech AV-710 - * Shuttle SN25P - * Onkyo SE-90PCI - * Onkyo SE-200PCI - * ESI Juli@ - * ESI Maya44 - * Hercules Fortissimo IV - * EGO-SYS WaveTerminal 192M - - model - Use the given board model, one of the following: - revo51, revo71, amp2000, prodigy71, prodigy71lt, - prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, - juli, aureon51, aureon71, universe, ap192, k8x800, - phase22, phase28, ms300, av710, se200pci, se90pci, - fortissimo4, sn25p, WT192M, maya44 - - This module supports multiple cards and autoprobe. - - Note: The supported board is detected by reading EEPROM or PCI - SSID (if EEPROM isn't available). You can override the - model by passing "model" module option in case that the - driver isn't configured properly or you want to try another - type for testing. - - Module snd-indigo - ----------------- - - Module for Echoaudio Indigo - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-indigodj - ------------------- - - Module for Echoaudio Indigo DJ - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-indigoio - ------------------- - - Module for Echoaudio Indigo IO - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-intel8x0 - ------------------- - - Module for AC'97 motherboards from Intel and compatibles. - * Intel i810/810E, i815, i820, i830, i84x, MX440 - ICH5, ICH6, ICH7, 6300ESB, ESB2 - * SiS 7012 (SiS 735) - * NVidia NForce, NForce2, NForce3, MCP04, CK804 - CK8, CK8S, MCP501 - * AMD AMD768, AMD8111 - * ALi m5455 - - ac97_clock - AC'97 codec clock base (0 = auto-detect) - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - buggy_irq - Enable workaround for buggy interrupts on some - motherboards (default yes on nForce chips, - otherwise off) - buggy_semaphore - Enable workaround for hardware with buggy - semaphores (e.g. on some ASUS laptops) - (default off) - spdif_aclink - Use S/PDIF over AC-link instead of direct connection - from the controller chip - (0 = off, 1 = on, -1 = default) - - This module supports one chip and autoprobe. - - Note: the latest driver supports auto-detection of chip clock. - if you still encounter too fast playback, specify the clock - explicitly via the module option "ac97_clock=41194". - - Joystick/MIDI ports are not supported by this driver. If your - motherboard has these devices, use the ns558 or snd-mpu401 - modules, respectively. - - The power-management is supported. - - Module snd-intel8x0m - -------------------- - - Module for Intel ICH (i8x0) chipset MC97 modems. - * Intel i810/810E, i815, i820, i830, i84x, MX440 - ICH5, ICH6, ICH7 - * SiS 7013 (SiS 735) - * NVidia NForce, NForce2, NForce2s, NForce3 - * AMD AMD8111 - * ALi m5455 - - ac97_clock - AC'97 codec clock base (0 = auto-detect) - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-interwave - -------------------- - - Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 - and other sound cards based on AMD InterWave (tm) chip. - - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) - pcm_voices - reserved PCM voices for the synthesizer (default 2) - effect - 1 = InterWave effects enable (default 0); - requires 8 voices - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) - - This module supports multiple cards, autoprobe and ISA PnP. - - Module snd-interwave-stb - ------------------------ - - Module for UltraSound 32-Pro (sound card from STB used by Compaq) - and other sound cards based on AMD InterWave (tm) chip with TEA6330T - circuit for extended control of bass, treble and master volume. - - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) - pcm_voices - reserved PCM voices for the synthesizer (default 2) - effect - 1 = InterWave effects enable (default 0); - requires 8 voices - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) - - This module supports multiple cards, autoprobe and ISA PnP. - - Module snd-jazz16 - ------------------- - - Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: - MVD1216 + MVA416 + MVA514. - - port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for SB DSP chip (3,5,7,9,10,15) - dma8 - DMA # for SB DSP chip (1,3) - dma16 - DMA # for SB DSP chip (5,7) - mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) - mpu_irq - MPU-401 irq # (2,3,5,7) - - This module supports multiple cards. - - Module snd-korg1212 - ------------------- - - Module for Korg 1212 IO PCI card - - This module supports multiple cards. - - Module snd-layla20 - ------------------ - - Module for Echoaudio Layla20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-layla24 - ------------------ - - Module for Echoaudio Layla24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-lola - --------------- - - Module for Digigram Lola PCI-e boards - - This module supports multiple cards. - - Module snd-lx6464es - ------------------- - - Module for Digigram LX6464ES boards - - This module supports multiple cards. - - Module snd-maestro3 - ------------------- - - Module for Allegro/Maestro3 chips - - external_amp - enable external amp (enabled by default) - amp_gpio - GPIO pin number for external amp (0-15) or - -1 for default pin (8 for allegro, 1 for - others) - - This module supports autoprobe and multiple chips. - - Note: the binding of amplifier is dependent on hardware. - If there is no sound even though all channels are unmuted, try to - specify other gpio connection via amp_gpio option. - For example, a Panasonic notebook might need "amp_gpio=0x0d" - option. - - The power-management is supported. - - Module snd-mia - --------------- - - Module for Echoaudio Mia - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-miro - --------------- - - Module for Miro soundcards: miroSOUND PCM 1 pro, - miroSOUND PCM 12, - miroSOUND PCM 20 Radio. - - port - Port # (0x530,0x604,0xe80,0xf40) - irq - IRQ # (5,7,9,10,11) - dma1 - 1st dma # (0,1,3) - dma2 - 2nd dma # (0,1) - mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) - mpu_irq - MPU-401 irq # (5,7,9,10) - fm_port - FM Port # (0x388) - wss - enable WSS mode - ide - enable onboard ide support - - Module snd-mixart - ----------------- - - Module for Digigram miXart8 sound cards. - - This module supports multiple cards. - Note: One miXart8 board will be represented as 4 alsa cards. - See MIXART.txt for details. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via mixartloader utility in alsa-tools package. - - Module snd-mona - --------------- - - Module for Echoaudio Mona - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-mpu401 - ----------------- - - Module for MPU-401 UART devices. - - port - port number or -1 (disable) - irq - IRQ number or -1 (disable) - pnp - PnP detection - 0 = disable, 1 = enable (default) - - This module supports multiple devices and PnP. - - Module snd-msnd-classic - ----------------------- - - Module for Turtle Beach MultiSound Classic, Tahiti or Monterey - soundcards. - - io - Port # for msnd-classic card - irq - IRQ # for msnd-classic card - mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, - 0xe0000 or 0xe8000) - write_ndelay - enable write ndelay (default = 1) - calibrate_signal - calibrate signal (default = 0) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - digital - Digital daughterboard present (default = 0) - cfg - Config port (0x250, 0x260 or 0x270) default = PnP - reset - Reset all devices - mpu_io - MPU401 I/O port - mpu_irq - MPU401 irq# - ide_io0 - IDE port #0 - ide_io1 - IDE port #1 - ide_irq - IDE irq# - joystick_io - Joystick I/O port - - The driver requires firmware files "turtlebeach/msndinit.bin" and - "turtlebeach/msndperm.bin" in the proper firmware directory. - - See Documentation/sound/oss/MultiSound for important information - about this driver. Note that it has been discontinued, but the - Voyetra Turtle Beach knowledge base entry for it is still available - at - http://www.turtlebeach.com - - Module snd-msnd-pinnacle - ------------------------ - - Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. - - io - Port # for pinnacle/fiji card - irq - IRQ # for pinnalce/fiji card - mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, - 0xe0000 or 0xe8000) - write_ndelay - enable write ndelay (default = 1) - calibrate_signal - calibrate signal (default = 0) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - The driver requires firmware files "turtlebeach/pndspini.bin" and - "turtlebeach/pndsperm.bin" in the proper firmware directory. - - Module snd-mtpav - ---------------- - - Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel - port). - - port - I/O port # for MTPAV (0x378,0x278, default=0x378) - irq - IRQ # for MTPAV (7,5, default=7) - hwports - number of supported hardware ports, default=8. - - Module supports only 1 card. This module has no enable option. - - Module snd-mts64 - ---------------- - - Module for Ego Systems (ESI) Miditerminal 4140 - - This module supports multiple devices. - Requires parport (CONFIG_PARPORT). - - Module snd-nm256 - ---------------- - - Module for NeoMagic NM256AV/ZX chips - - playback_bufsize - max playback frame size in kB (4-128kB) - capture_bufsize - max capture frame size in kB (4-128kB) - force_ac97 - 0 or 1 (disabled by default) - buffer_top - specify buffer top address - use_cache - 0 or 1 (disabled by default) - vaio_hack - alias buffer_top=0x25a800 - reset_workaround - enable AC97 RESET workaround for some laptops - reset_workaround2 - enable extended AC97 RESET workaround for some - other laptops - - This module supports one chip and autoprobe. - - The power-management is supported. - - Note: on some notebooks the buffer address cannot be detected - automatically, or causes hang-up during initialization. - In such a case, specify the buffer top address explicitly via - the buffer_top option. - For example, - Sony F250: buffer_top=0x25a800 - Sony F270: buffer_top=0x272800 - The driver supports only ac97 codec. It's possible to force - to initialize/use ac97 although it's not detected. In such a - case, use force_ac97=1 option - but *NO* guarantee whether it - works! - - Note: The NM256 chip can be linked internally with non-AC97 - codecs. This driver supports only the AC97 codec, and won't work - with machines with other (most likely CS423x or OPL3SAx) chips, - even though the device is detected in lspci. In such a case, try - other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP - but some doesn't have ISA PnP. You'll need to specify isapnp=0 - and proper hardware parameters in the case without ISA PnP. - - Note: some laptops need a workaround for AC97 RESET. For the - known hardware like Dell Latitude LS and Sony PCG-F305, this - workaround is enabled automatically. For other laptops with a - hard freeze, you can try reset_workaround=1 option. - - Note: Dell Latitude CSx laptops have another problem regarding - AC97 RESET. On these laptops, reset_workaround2 option is - turned on as default. This option is worth to try if the - previous reset_workaround option doesn't help. - - Note: This driver is really crappy. It's a porting from the - OSS driver, which is a result of black-magic reverse engineering. - The detection of codec will fail if the driver is loaded *after* - X-server as described above. You might be able to force to load - the module, but it may result in hang-up. Hence, make sure that - you load this module *before* X if you encounter this kind of - problem. - - Module snd-opl3sa2 - ------------------ - - Module for Yamaha OPL3-SA2/SA3 sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - control port # for OPL3-SA chip (0x370) - sb_port - SB port # for OPL3-SA chip (0x220,0x240) - wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) - midi_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - fm_port - FM port # for OPL3-SA chip (0x388), -1 = disable - irq - IRQ # for OPL3-SA chip (5,7,9,10) - dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3) - dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable - - This module supports multiple cards and ISA PnP. It does not support - autoprobe (if ISA PnP is not used) thus all ports must be specified!!! - - The power-management is supported. - - Module snd-opti92x-ad1848 - ------------------------- - - Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. - Module works with OAK Mozart cards as well. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-opti92x-cs4231 - ------------------------- - - Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - dma2 - second DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-opti93x - ------------------ - - Module for sound cards based on OPTi 82c93x chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - dma2 - second DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-oxygen - ----------------- - - Module for sound cards based on the C-Media CMI8786/8787/8788 chip: - * Asound A-8788 - * Asus Xonar DG/DGX - * AuzenTech X-Meridian - * AuzenTech X-Meridian 2G - * Bgears b-Enspirer - * Club3D Theatron DTS - * HT-Omega Claro (plus) - * HT-Omega Claro halo (XT) - * Kuroutoshikou CMI8787-HG2PCI - * Razer Barracuda AC-1 - * Sondigo Inferno - * TempoTec HiFier Fantasia - * TempoTec HiFier Serenade - - This module supports autoprobe and multiple cards. - - Module snd-pcsp - ----------------- - - Module for internal PC-Speaker. - - nopcm - Disable PC-Speaker PCM sound. Only beeps remain. - nforce_wa - enable NForce chipset workaround. Expect bad sound. - - This module supports system beeps, some kind of PCM playback and - even a few mixer controls. - - Module snd-pcxhr - ---------------- - - Module for Digigram PCXHR boards - - This module supports multiple cards. - - Module snd-portman2x4 - --------------------- - - Module for Midiman Portman 2x4 parallel port MIDI interface - - This module supports multiple cards. - - Module snd-powermac (on ppc only) - --------------------------------- - - Module for PowerMac, iMac and iBook on-board soundchips - - enable_beep - enable beep using PCM (enabled as default) - - Module supports autoprobe a chip. - - Note: the driver may have problems regarding endianness. - - The power-management is supported. - - Module snd-pxa2xx-ac97 (on arm only) - ------------------------------------ - - Module for AC97 driver for the Intel PXA2xx chip - - For ARM architecture only. - - The power-management is supported. - - Module snd-riptide - ------------------ - - Module for Conexant Riptide chip - - joystick_port - Joystick port # (default: 0x200) - mpu_port - MPU401 port # (default: 0x330) - opl3_port - OPL3 port # (default: 0x388) - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - You need to install the firmware file "riptide.hex" to the standard - firmware path (e.g. /lib/firmware). - - Module snd-rme32 - ---------------- - - Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, - Prodif96 and Prodif Gold) sound cards. - - This module supports multiple cards. - - Module snd-rme96 - ---------------- - - Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. - - This module supports multiple cards. - - Module snd-rme9652 - ------------------ - - Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. - - precise_ptr - Enable precise pointer (doesn't work reliably). - (default = 0) - - This module supports multiple cards. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any RME9652 cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-sa11xx-uda1341 (on arm only) - --------------------------------------- - - Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. - - Module supports only one card. - Module has no enable and index options. - - The power-management is supported. - - Module snd-sb8 - -------------- - - Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, - SoundBlaster 2.0, - SoundBlaster Pro - - port - port # for SB DSP chip (0x220,0x240,0x260) - irq - IRQ # for SB DSP chip (5,7,9,10) - dma8 - DMA # for SB DSP chip (1,3) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-sb16 and snd-sbawe - ----------------------------- - - Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), - SoundBlaster AWE 32 (PnP), - SoundBlaster AWE 64 PnP - - mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) - csp - ASP/CSP chip support - 0 = disable (default), 1 = enable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for SB DSP 4.x chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) - (snd-sbawe module only) - irq - IRQ # for SB DSP 4.x chip (5,7,9,10) - dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) - dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) - - This module supports multiple cards, autoprobe and ISA PnP. - - Note: To use Vibra16X cards in 16-bit half duplex mode, you must - disable 16bit DMA with dma16 = -1 module parameter. - Also, all Sound Blaster 16 type cards can operate in 16-bit - half duplex mode through 8-bit DMA channel by disabling their - 16-bit DMA channel. - - The power-management is supported. - - Module snd-sc6000 - ----------------- - - Module for Gallant SC-6000 soundcard and later models: SC-6600 - and SC-7000. - - port - Port # (0x220 or 0x240) - mss_port - MSS Port # (0x530 or 0xe80) - irq - IRQ # (5,7,9,10,11) - mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq - dma - DMA # (1,3,0) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - - Module snd-sscape - ----------------- - - Module for ENSONIQ SoundScape cards. - - port - Port # (PnP setup) - wss_port - WSS Port # (PnP setup) - irq - IRQ # (PnP setup) - mpu_irq - MPU-401 IRQ # (PnP setup) - dma - DMA # (PnP setup) - dma2 - 2nd DMA # (PnP setup, -1 to disable) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - The driver requires the firmware loader support on kernel. - - Module snd-sun-amd7930 (on sparc only) - -------------------------------------- - - Module for AMD7930 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-cs4231 (on sparc only) - ------------------------------------- - - Module for CS4231 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-dbri (on sparc only) - ----------------------------------- - - Module for DBRI sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-wavefront - -------------------- - - Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. - - use_cs4232_midi - Use CS4232 MPU-401 interface - (inaccessibly located inside your computer) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - cs4232_pcm_port - Port # for CS4232 PCM interface. - cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). - cs4232_mpu_port - Port # for CS4232 MPU-401 interface. - cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). - ics2115_port - Port # for ICS2115 - ics2115_irq - IRQ # for ICS2115 - fm_port - FM OPL-3 Port # - dma1 - DMA1 # for CS4232 PCM interface. - dma2 - DMA2 # for CS4232 PCM interface. - - The below are options for wavefront_synth features: - wf_raw - Assume that we need to boot the OS (default:no) - If yes, then during driver loading, the state of the board is - ignored, and we reset the board and load the firmware anyway. - fx_raw - Assume that the FX process needs help (default:yes) - If false, we'll leave the FX processor in whatever state it is - when the driver is loaded. The default is to download the - microprogram and associated coefficients to set it up for - "default" operation, whatever that means. - debug_default - Debug parameters for card initialization - wait_usecs - How long to wait without sleeping, usecs - (default:150) - This magic number seems to give pretty optimal throughput - based on my limited experimentation. - If you want to play around with it and find a better value, be - my guest. Remember, the idea is to get a number that causes us - to just busy wait for as many WaveFront commands as possible, - without coming up with a number so large that we hog the whole - CPU. - Specifically, with this number, out of about 134,000 status - waits, only about 250 result in a sleep. - sleep_interval - How long to sleep when waiting for reply - (default: 100) - sleep_tries - How many times to try sleeping during a wait - (default: 50) - ospath - Pathname to processed ICS2115 OS firmware - (default:wavefront.os) - The path name of the ISC2115 OS firmware. In the recent - version, it's handled via firmware loader framework, so it - must be installed in the proper path, typically, - /lib/firmware. - reset_time - How long to wait for a reset to take effect - (default:2) - ramcheck_time - How many seconds to wait for the RAM test - (default:20) - osrun_time - How many seconds to wait for the ICS2115 OS - (default:10) - - This module supports multiple cards and ISA PnP. - - Note: the firmware file "wavefront.os" was located in the earlier - version in /etc. Now it's loaded via firmware loader, and - must be in the proper firmware path, such as /lib/firmware. - Copy (or symlink) the file appropriately if you get an error - regarding firmware downloading after upgrading the kernel. - - Module snd-sonicvibes - --------------------- - - Module for S3 SonicVibes PCI sound cards. - * PINE Schubert 32 PCI - - reverb - Reverb Enable - 1 = enable, 0 = disable (default) - - SoundCard must have onboard SRAM for this. - mge - Mic Gain Enable - 1 = enable, 0 = disable (default) - - This module supports multiple cards and autoprobe. - - Module snd-serial-u16550 - ------------------------ - - Module for UART16550A serial MIDI ports. - - port - port # for UART16550A chip - irq - IRQ # for UART16550A chip, -1 = poll mode - speed - speed in bauds (9600,19200,38400,57600,115200) - 38400 = default - base - base for divisor in bauds (57600,115200,230400,460800) - 115200 = default - outs - number of MIDI ports in a serial port (1-4) - 1 = default - adaptor - Type of adaptor. - 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, - 3 = MS-124W M/B, 4 = Generic - - This module supports multiple cards. This module does not support autoprobe - thus the main port must be specified!!! Other options are optional. - - Module snd-trident - ------------------ - - Module for Trident 4DWave DX/NX sound cards. - * Best Union Miss Melody 4DWave PCI - * HIS 4DWave PCI - * Warpspeed ONSpeed 4DWave PCI - * AzTech PCI 64-Q3D - * Addonics SV 750 - * CHIC True Sound 4Dwave - * Shark Predator4D-PCI - * Jaton SonicWave 4D - * SiS SI7018 PCI Audio - * Hoontech SoundTrack Digital 4DWave NX - - pcm_channels - max channels (voices) reserved for PCM - wavetable_size - max wavetable size in kB (4-?kb) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-ua101 - ---------------- - - Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. - - This module supports multiple devices, autoprobe and hotplugging. - - Module snd-usb-audio - -------------------- - - Module for USB audio and USB MIDI devices. - - vid - Vendor ID for the device (optional) - pid - Product ID for the device (optional) - nrpacks - Max. number of packets per URB (default: 8) - device_setup - Device specific magic number (optional) - - Influence depends on the device - - Default: 0x0000 - ignore_ctl_error - Ignore any USB-controller regarding mixer - interface (default: no) - autoclock - Enable auto-clock selection for UAC2 devices - (default: yes) - quirk_alias - Quirk alias list, pass strings like - "0123abcd:5678beef", which applies the existing - quirk for the device 5678:beef to a new device - 0123:abcd. - - This module supports multiple devices, autoprobe and hotplugging. - - NB: nrpacks parameter can be modified dynamically via sysfs. - Don't put the value over 20. Changing via sysfs has no sanity - check. - NB: ignore_ctl_error=1 may help when you get an error at accessing - the mixer element such as URB error -22. This happens on some - buggy USB device or the controller. - NB: quirk_alias option is provided only for testing / development. - If you want to have a proper support, contact to upstream for - adding the matching quirk in the driver code statically. - - Module snd-usb-caiaq - -------------------- - - Module for caiaq UB audio interfaces, - * Native Instruments RigKontrol2 - * Native Instruments Kore Controller - * Native Instruments Audio Kontrol 1 - * Native Instruments Audio 8 DJ - - This module supports multiple devices, autoprobe and hotplugging. - - Module snd-usb-usx2y - -------------------- - - Module for Tascam USB US-122, US-224 and US-428 devices. - - This module supports multiple devices, autoprobe and hotplugging. - - Note: you need to load the firmware via usx2yloader utility included - in alsa-tools and alsa-firmware packages. - - Module snd-via82xx - ------------------ - - Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, - 8233A, 8233C, 8235, 8237 (south) bridge. - - mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup - [VIA686A/686B only] - joystick - Enable joystick (default off) [VIA686A/686B only] - ac97_clock - AC'97 codec clock base (default 48000Hz) - dxs_support - support DXS channels, - 0 = auto (default), 1 = enable, 2 = disable, - 3 = 48k only, 4 = no VRA, 5 = enable any sample - rate and different sample rates on different - channels - [VIA8233/C, 8235, 8237 only] - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - - This module supports one chip and autoprobe. - - Note: on some SMP motherboards like MSI 694D the interrupts might - not be generated properly. In such a case, please try to - set the SMP (or MPS) version on BIOS to 1.1 instead of - default value 1.4. Then the interrupt number will be - assigned under 15. You might also upgrade your BIOS. - - Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) - channels as the first PCM. On these channels, up to 4 - streams can be played at the same time, and the controller - can perform sample rate conversion with separate rates for - each channel. - As default (dxs_support = 0), 48k fixed rate is chosen - except for the known devices since the output is often - noisy except for 48k on some mother boards due to the - bug of BIOS. - Please try once dxs_support=5 and if it works on other - sample rates (e.g. 44.1kHz of mp3 playback), please let us - know the PCI subsystem vendor/device id's (output of - "lspci -nv"). - If dxs_support=5 does not work, try dxs_support=4; if it - doesn't work too, try dxs_support=1. (dxs_support=1 is - usually for old motherboards. The correct implemented - board should work with 4 or 5.) If it still doesn't - work and the default setting is ok, dxs_support=3 is the - right choice. If the default setting doesn't work at all, - try dxs_support=2 to disable the DXS channels. - In any cases, please let us know the result and the - subsystem vendor/device ids. See "Links and Addresses" - below. - - Note: for the MPU401 on VIA823x, use snd-mpu401 driver - additionally. The mpu_port option is for VIA686 chips only. - - The power-management is supported. - - Module snd-via82xx-modem - ------------------------ - - Module for VIA82xx AC97 modem - - ac97_clock - AC'97 codec clock base (default 48000Hz) - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-virmidi - ------------------ - - Module for virtual rawmidi devices. - This module creates virtual rawmidi devices which communicate - to the corresponding ALSA sequencer ports. - - midi_devs - MIDI devices # (1-4, default=4) - - This module supports multiple cards. - - Module snd-virtuoso - ------------------- - - Module for sound cards based on the Asus AV66/AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), - Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. - - This module supports autoprobe and multiple cards. - - Module snd-vx222 - ---------------- - - Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. - - mic - Enable Microphone on V222 Mic (NYI) - ibl - Capture IBL size. (default = 0, minimum size) - - This module supports multiple cards. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via vxloader utility in alsa-tools package. To invoke - vxloader automatically, add the following to /etc/modprobe.d/alsa.conf - - install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader - - (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to - /etc/modules.conf, instead.) - IBL size defines the interrupts period for PCM. The smaller size - gives smaller latency but leads to more CPU consumption, too. - The size is usually aligned to 126. As default (=0), the smallest - size is chosen. The possible IBL values can be found in - /proc/asound/cardX/vx-status proc file. - - The power-management is supported. - - Module snd-vxpocket - ------------------- - - Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. - - ibl - Capture IBL size. (default = 0, minimum size) - - This module supports multiple cards. The module is compiled only when - PCMCIA is supported on kernel. - - With the older 2.6.x kernel, to activate the driver via the card - manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the - sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no - longer require a config file. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via vxloader utility in alsa-tools package. - - About capture IBL, see the description of snd-vx222 module. - - Note: snd-vxp440 driver is merged to snd-vxpocket driver since - ALSA 1.0.10. - - The power-management is supported. - - Module snd-ymfpci - ----------------- - - Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). - - mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default, - 1 (auto-detect for YMF744/754 only) - fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default - 1 (auto-detect for YMF744/754 only) - joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default, - 1 (auto-detect) - rear_switch - enable shared rear/line-in switch (bool) - - This module supports autoprobe and multiple chips. - - The power-management is supported. - - Module snd-pdaudiocf - -------------------- - - Module for Sound Core PDAudioCF sound card. - - The power-management is supported. - - -AC97 Quirk Option -================= - -The ac97_quirk option is used to enable/override the workaround for -specific devices on drivers for on-board AC'97 controllers like -snd-intel8x0. Some hardware have swapped output pins between Master -and Headphone, or Surround (thanks to confusion of AC'97 -specifications from version to version :-) - -The driver provides the auto-detection of known problematic devices, -but some might be unknown or wrongly detected. In such a case, pass -the proper value with this option. - -The following strings are accepted: - - default Don't override the default setting - - none Disable the quirk - - hp_only Bind Master and Headphone controls as a single control - - swap_hp Swap headphone and master controls - - swap_surround Swap master and surround controls - - ad_sharing For AD1985, turn on OMS bit and use headphone - - alc_jack For ALC65x, turn on the jack sense mode - - inv_eapd Inverted EAPD implementation - - mute_led Bind EAPD bit for turning on/off mute LED - -For backward compatibility, the corresponding integer value -1, 0, -... are accepted, too. - -For example, if "Master" volume control has no effect on your device -but only "Headphone" does, pass ac97_quirk=hp_only module option. - - -Configuring Non-ISAPNP Cards -============================ - -When the kernel is configured with ISA-PnP support, the modules -supporting the isapnp cards will have module options "isapnp". -If this option is set, *only* the ISA-PnP devices will be probed. -For probing the non ISA-PnP cards, you have to pass "isapnp=0" option -together with the proper i/o and irq configuration. - -When the kernel is configured without ISA-PnP support, isapnp option -will be not built in. - - -Module Autoloading Support -========================== - -The ALSA drivers can be loaded automatically on demand by defining -module aliases. The string 'snd-card-%1' is requested for ALSA native -devices where %i is sound card number from zero to seven. - -To auto-load an ALSA driver for OSS services, define the string -'sound-slot-%i' where %i means the slot number for OSS, which -corresponds to the card index of ALSA. Usually, define this -as the same card module. - -An example configuration for a single emu10k1 card is like below: ------ /etc/modprobe.d/alsa.conf -alias snd-card-0 snd-emu10k1 -alias sound-slot-0 snd-emu10k1 ------ /etc/modprobe.d/alsa.conf - -The available number of auto-loaded sound cards depends on the module -option "cards_limit" of snd module. As default it's set to 1. -To enable the auto-loading of multiple cards, specify the number of -sound cards in that option. - -When multiple cards are available, it'd better to specify the index -number for each card via module option, too, so that the order of -cards is kept consistent. - -An example configuration for two sound cards is like below: - ------ /etc/modprobe.d/alsa.conf -# ALSA portion -options snd cards_limit=2 -alias snd-card-0 snd-interwave -alias snd-card-1 snd-ens1371 -options snd-interwave index=0 -options snd-ens1371 index=1 -# OSS/Free portion -alias sound-slot-0 snd-interwave -alias sound-slot-1 snd-ens1371 ------ /etc/modprobe.d/alsa.conf - -In this example, the interwave card is always loaded as the first card -(index 0) and ens1371 as the second (index 1). - -Alternative (and new) way to fixate the slot assignment is to use -"slots" option of snd module. In the case above, specify like the -following: - -options snd slots=snd-interwave,snd-ens1371 - -Then, the first slot (#0) is reserved for snd-interwave driver, and -the second (#1) for snd-ens1371. You can omit index option in each -driver if slots option is used (although you can still have them at -the same time as long as they don't conflict). - -The slots option is especially useful for avoiding the possible -hot-plugging and the resultant slot conflict. For example, in the -case above again, the first two slots are already reserved. If any -other driver (e.g. snd-usb-audio) is loaded before snd-interwave or -snd-ens1371, it will be assigned to the third or later slot. - -When a module name is given with '!', the slot will be given for any -modules but that name. For example, "slots=!snd-pcsp" will reserve -the first slot for any modules but snd-pcsp. - - -ALSA PCM devices to OSS devices mapping -======================================= - -/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 -/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 -/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 -/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 -/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 -/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 -/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 -/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 -/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 - -The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means -sound card number and second means device number. The ALSA devices -have either 'c' or 'p' suffix indicating the direction, capture and -playback, respectively. - -Please note that the device mapping above may be varied via the module -options of snd-pcm-oss module. - - -Proc interfaces (/proc/asound) -============================== - -/proc/asound/card#/pcm#[cp]/oss -------------------------------- - String "erase" - erase all additional information about OSS applications - String "<app_name> <fragments> <fragment_size> [<options>]" - - <app_name> - name of application with (higher priority) or without path - <fragments> - number of fragments or zero if auto - <fragment_size> - size of fragment in bytes or zero if auto - <options> - optional parameters - - disable the application tries to open a pcm device for - this channel but does not want to use it. - (Cause a bug or mmap needs) - It's good for Quake etc... - - direct don't use plugins - - block force block mode (rvplayer) - - non-block force non-block mode - - whole-frag write only whole fragments (optimization affecting - playback only) - - no-silence do not fill silence ahead to avoid clicks - - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl - instead of filled blocks - - Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss - echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss - echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss - - -Early Buffer Allocation -======================= - -Some drivers (e.g. hdsp) require the large contiguous buffers, and -sometimes it's too late to find such spaces when the driver module is -actually loaded due to memory fragmentation. You can pre-allocate the -PCM buffers by loading snd-page-alloc module and write commands to its -proc file in prior, for example, in the early boot stage like -/etc/init.d/*.local scripts. - -Reading the proc file /proc/drivers/snd-page-alloc shows the current -usage of page allocation. In writing, you can send the following -commands to the snd-page-alloc driver: - - - add VENDOR DEVICE MASK SIZE BUFFERS - - VENDOR and DEVICE are PCI vendor and device IDs. They take - integer numbers (0x prefix is needed for the hex). - MASK is the PCI DMA mask. Pass 0 if not restricted. - SIZE is the size of each buffer to allocate. You can pass - k and m suffix for KB and MB. The max number is 16MB. - BUFFERS is the number of buffers to allocate. It must be greater - than 0. The max number is 4. - - - erase - - This will erase the all pre-allocated buffers which are not in - use. - - -Links and Addresses -=================== - - ALSA project homepage - http://www.alsa-project.org - - Kernel Bugzilla - http://bugzilla.kernel.org/ - - ALSA Developers ML - mailto:alsa-devel@alsa-project.org - - alsa-info.sh script - http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt deleted file mode 100644 index 3fc1cf50d28e..000000000000 --- a/Documentation/sound/alsa/ControlNames.txt +++ /dev/null @@ -1,107 +0,0 @@ -This document describes standard names of mixer controls. - -Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION - -DIRECTION: - <nothing> (both directions) - Playback - Capture - Bypass Playback - Bypass Capture - -FUNCTION: - Switch (on/off switch) - Volume - Route (route control, hardware specific) - -CHANNEL: - <nothing> (channel independent, or applies to all channels) - Front - Surround (rear left/right in 4.0/5.1 surround) - CLFE - Center - LFE - Side (side left/right for 7.1 surround) - -LOCATION: (physical location of source) - Front - Rear - Dock (docking station) - Internal - -SOURCE: - Master - Master Mono - Hardware Master - Speaker (internal speaker) - Bass Speaker (internal LFE speaker) - Headphone - Line Out - Beep (beep generator) - Phone - Phone Input - Phone Output - Synth - FM - Mic - Headset Mic (mic part of combined headset jack - 4-pin headphone + mic) - Headphone Mic (mic part of either/or - 3-pin headphone or mic) - Line (input only, use "Line Out" for output) - CD - Video - Zoom Video - Aux - PCM - PCM Pan - Loopback - Analog Loopback (D/A -> A/D loopback) - Digital Loopback (playback -> capture loopback - without analog path) - Mono - Mono Output - Multi - ADC - Wave - Music - I2S - IEC958 - HDMI - SPDIF (output only) - SPDIF In - Digital In - HDMI/DP (either HDMI or DisplayPort) - -Exceptions (deprecated): - [Analogue|Digital] Capture Source - [Analogue|Digital] Capture Switch (aka input gain switch) - [Analogue|Digital] Capture Volume (aka input gain volume) - [Analogue|Digital] Playback Switch (aka output gain switch) - [Analogue|Digital] Playback Volume (aka output gain volume) - Tone Control - Switch - Tone Control - Bass - Tone Control - Treble - 3D Control - Switch - 3D Control - Center - 3D Control - Depth - 3D Control - Wide - 3D Control - Space - 3D Control - Level - Mic Boost [(?dB)] - -PCM interface: - - Sample Clock Source { "Word", "Internal", "AutoSync" } - Clock Sync Status { "Lock", "Sync", "No Lock" } - External Rate /* external capture rate */ - Capture Rate /* capture rate taken from external source */ - -IEC958 (S/PDIF) interface: - - IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */ - IEC958 [...] [Playback|Capture] Volume /* digital volume control */ - IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */ - IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */ - IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */ - IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */ - IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */ - IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */ - IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */ diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt deleted file mode 100644 index ec099d4343f2..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ /dev/null @@ -1,324 +0,0 @@ - Model name Description - ---------- ----------- -ALC880 -====== - 3stack 3-jack in back and a headphone out - 3stack-digout 3-jack in back, a HP out and a SPDIF out - 5stack 5-jack in back, 2-jack in front - 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out - 6stack 6-jack in back, 2-jack in front - 6stack-digout 6-jack with a SPDIF out - -ALC260 -====== - gpio1 Enable GPIO1 - coef Enable EAPD via COEF table - fujitsu Quirk for FSC S7020 - fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support - -ALC262 -====== - inv-dmic Inverted internal mic workaround - -ALC267/268 -========== - inv-dmic Inverted internal mic workaround - hp-eapd Disable HP EAPD on NID 0x15 - -ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) -====== - laptop-amic Laptops with analog-mic input - laptop-dmic Laptops with digital-mic input - alc269-dmic Enable ALC269(VA) digital mic workaround - alc271-dmic Enable ALC271X digital mic workaround - inv-dmic Inverted internal mic workaround - headset-mic Indicates a combined headset (headphone+mic) jack - headset-mode More comprehensive headset support for ALC269 & co - headset-mode-no-hp-mic Headset mode support without headphone mic - lenovo-dock Enables docking station I/O for some Lenovos - hp-gpio-led GPIO LED support on HP laptops - dell-headset-multi Headset jack, which can also be used as mic-in - dell-headset-dock Headset jack (without mic-in), and also dock I/O - alc283-dac-wcaps Fixups for Chromebook with ALC283 - alc283-sense-combo Combo jack sensing on ALC283 - tpt440-dock Pin configs for Lenovo Thinkpad Dock support - -ALC66x/67x/892 -============== - mario Chromebook mario model fixup - asus-mode1 ASUS - asus-mode2 ASUS - asus-mode3 ASUS - asus-mode4 ASUS - asus-mode5 ASUS - asus-mode6 ASUS - asus-mode7 ASUS - asus-mode8 ASUS - inv-dmic Inverted internal mic workaround - dell-headset-multi Headset jack, which can also be used as mic-in - -ALC680 -====== - N/A - -ALC88x/898/1150 -====================== - acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G - acer-aspire-8930g Acer Aspire 8330G/6935G - acer-aspire Acer Aspire others - inv-dmic Inverted internal mic workaround - no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) - -ALC861/660 -========== - N/A - -ALC861VD/660VD -============== - N/A - -CMI9880 -======= - minimal 3-jack in back - min_fp 3-jack in back, 2-jack in front - full 6-jack in back, 2-jack in front - full_dig 6-jack in back, 2-jack in front, SPDIF I/O - allout 5-jack in back, 2-jack in front, SPDIF out - auto auto-config reading BIOS (default) - -AD1882 / AD1882A -================ - 3stack 3-stack mode - 3stack-automute 3-stack with automute front HP (default) - 6stack 6-stack mode - -AD1884A / AD1883 / AD1984A / AD1984B -==================================== - desktop 3-stack desktop (default) - laptop laptop with HP jack sensing - mobile mobile devices with HP jack sensing - thinkpad Lenovo Thinkpad X300 - touchsmart HP Touchsmart - -AD1884 -====== - N/A - -AD1981 -====== - basic 3-jack (default) - hp HP nx6320 - thinkpad Lenovo Thinkpad T60/X60/Z60 - toshiba Toshiba U205 - -AD1983 -====== - N/A - -AD1984 -====== - basic default configuration - thinkpad Lenovo Thinkpad T61/X61 - dell_desktop Dell T3400 - -AD1986A -======= - 3stack 3-stack, shared surrounds - laptop 2-channel only (FSC V2060, Samsung M50) - laptop-imic 2-channel with built-in mic - eapd Turn on EAPD constantly - -AD1988/AD1988B/AD1989A/AD1989B -============================== - 6stack 6-jack - 6stack-dig ditto with SPDIF - 3stack 3-jack - 3stack-dig ditto with SPDIF - laptop 3-jack with hp-jack automute - laptop-dig ditto with SPDIF - auto auto-config reading BIOS (default) - -Conexant 5045 -============= - laptop-hpsense Laptop with HP sense (old model laptop) - laptop-micsense Laptop with Mic sense (old model fujitsu) - laptop-hpmicsense Laptop with HP and Mic senses - benq Benq R55E - laptop-hp530 HP 530 laptop - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5047 -============= - laptop Basic Laptop config - laptop-hp Laptop config for some HP models (subdevice 30A5) - laptop-eapd Laptop config with EAPD support - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5051 -============= - laptop Basic Laptop config (default) - hp HP Spartan laptop - hp-dv6736 HP dv6736 - hp-f700 HP Compaq Presario F700 - ideapad Lenovo IdeaPad laptop - toshiba Toshiba Satellite M300 - -Conexant 5066 -============= - laptop Basic Laptop config (default) - hp-laptop HP laptops, e g G60 - asus Asus K52JU, Lenovo G560 - dell-laptop Dell laptops - dell-vostro Dell Vostro - olpc-xo-1_5 OLPC XO 1.5 - ideapad Lenovo IdeaPad U150 - thinkpad Lenovo Thinkpad - -STAC9200 -======== - ref Reference board - oqo OQO Model 2 - dell-d21 Dell (unknown) - dell-d22 Dell (unknown) - dell-d23 Dell (unknown) - dell-m21 Dell Inspiron 630m, Dell Inspiron 640m - dell-m22 Dell Latitude D620, Dell Latitude D820 - dell-m23 Dell XPS M1710, Dell Precision M90 - dell-m24 Dell Latitude 120L - dell-m25 Dell Inspiron E1505n - dell-m26 Dell Inspiron 1501 - dell-m27 Dell Inspiron E1705/9400 - gateway-m4 Gateway laptops with EAPD control - gateway-m4-2 Gateway laptops with EAPD control - panasonic Panasonic CF-74 - auto BIOS setup (default) - -STAC9205/9254 -============= - ref Reference board - dell-m42 Dell (unknown) - dell-m43 Dell Precision - dell-m44 Dell Inspiron - eapd Keep EAPD on (e.g. Gateway T1616) - auto BIOS setup (default) - -STAC9220/9221 -============= - ref Reference board - 3stack D945 3stack - 5stack D945 5stack + SPDIF - intel-mac-v1 Intel Mac Type 1 - intel-mac-v2 Intel Mac Type 2 - intel-mac-v3 Intel Mac Type 3 - intel-mac-v4 Intel Mac Type 4 - intel-mac-v5 Intel Mac Type 5 - intel-mac-auto Intel Mac (detect type according to subsystem id) - macmini Intel Mac Mini (equivalent with type 3) - macbook Intel Mac Book (eq. type 5) - macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) - macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) - imac-intel Intel iMac (eq. type 2) - imac-intel-20 Intel iMac (newer version) (eq. type 3) - ecs202 ECS/PC chips - dell-d81 Dell (unknown) - dell-d82 Dell (unknown) - dell-m81 Dell (unknown) - dell-m82 Dell XPS M1210 - auto BIOS setup (default) - -STAC9202/9250/9251 -================== - ref Reference board, base config - m1 Some Gateway MX series laptops (NX560XL) - m1-2 Some Gateway MX series laptops (MX6453) - m2 Some Gateway MX series laptops (M255) - m2-2 Some Gateway MX series laptops - m3 Some Gateway MX series laptops - m5 Some Gateway MX series laptops (MP6954) - m6 Some Gateway NX series laptops - auto BIOS setup (default) - -STAC9227/9228/9229/927x -======================= - ref Reference board - ref-no-jd Reference board without HP/Mic jack detection - 3stack D965 3stack - 5stack D965 5stack + SPDIF - 5stack-no-fp D965 5stack without front panel - dell-3stack Dell Dimension E520 - dell-bios Fixes with Dell BIOS setup - dell-bios-amic Fixes with Dell BIOS setup including analog mic - volknob Fixes with volume-knob widget 0x24 - auto BIOS setup (default) - -STAC92HD71B* -============ - ref Reference board - dell-m4-1 Dell desktops - dell-m4-2 Dell desktops - dell-m4-3 Dell desktops - hp-m4 HP mini 1000 - hp-dv5 HP dv series - hp-hdx HP HDX series - hp-dv4-1222nr HP dv4-1222nr (with LED support) - auto BIOS setup (default) - -STAC92HD73* -=========== - ref Reference board - no-jd BIOS setup but without jack-detection - intel Intel DG45* mobos - dell-m6-amic Dell desktops/laptops with analog mics - dell-m6-dmic Dell desktops/laptops with digital mics - dell-m6 Dell desktops/laptops with both type of mics - dell-eq Dell desktops/laptops - alienware Alienware M17x - auto BIOS setup (default) - -STAC92HD83* -=========== - ref Reference board - mic-ref Reference board with power management for ports - dell-s14 Dell laptop - dell-vostro-3500 Dell Vostro 3500 laptop - hp-dv7-4000 HP dv-7 4000 - hp_cNB11_intquad HP CNB models with 4 speakers - hp-zephyr HP Zephyr - hp-led HP with broken BIOS for mute LED - hp-inv-led HP with broken BIOS for inverted mute LED - hp-mic-led HP with mic-mute LED - headset-jack Dell Latitude with a 4-pin headset jack - hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) - hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) - hp-bnb13-eq Hardware equalizer setup for HP laptops - auto BIOS setup (default) - -STAC92HD95 -========== - hp-led LED support for HP laptops - hp-bass Bass HPF setup for HP Spectre 13 - -STAC9872 -======== - vaio VAIO laptop without SPDIF - auto BIOS setup (default) - -Cirrus Logic CS4206/4207 -======================== - mbp55 MacBook Pro 5,5 - imac27 IMac 27 Inch - auto BIOS setup (default) - -Cirrus Logic CS4208 -=================== - mba6 MacBook Air 6,1 and 6,2 - gpio0 Enable GPIO 0 amp - auto BIOS setup (default) - -VIA VT17xx/VT18xx/VT20xx -======================== - auto BIOS setup (default) diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt deleted file mode 100644 index 1b0ac06ba95d..000000000000 --- a/Documentation/sound/alsa/VIA82xx-mixer.txt +++ /dev/null @@ -1,8 +0,0 @@ - - VIA82xx mixer - ============= - -On many VIA82xx boards, the 'Input Source Select' mixer control does not work. -Setting it to 'Input2' on such boards will cause recording to hang, or fail -with EIO (input/output error) via OSS emulation. This control should be left -at 'Input1' for such cards. diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt deleted file mode 100644 index 72eced86f035..000000000000 --- a/Documentation/sound/alsa/alsa-parameters.txt +++ /dev/null @@ -1,135 +0,0 @@ - ALSA Kernel Parameters - ~~~~~~~~~~~~~~~~~~~~~~ - -See Documentation/admin-guide/kernel-parameters.rst for general information on -specifying module parameters. - -This document may not be entirely up to date and comprehensive. The command -"modinfo -p ${modulename}" shows a current list of all parameters of a loadable -module. Loadable modules, after being loaded into the running kernel, also -reveal their parameters in /sys/module/${modulename}/parameters/. Some of these -parameters may be changed at runtime by the command -"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". - - - snd-ad1816a= [HW,ALSA] - - snd-ad1848= [HW,ALSA] - - snd-ali5451= [HW,ALSA] - - snd-als100= [HW,ALSA] - - snd-als4000= [HW,ALSA] - - snd-azt2320= [HW,ALSA] - - snd-cmi8330= [HW,ALSA] - - snd-cmipci= [HW,ALSA] - - snd-cs4231= [HW,ALSA] - - snd-cs4232= [HW,ALSA] - - snd-cs4236= [HW,ALSA] - - snd-cs4281= [HW,ALSA] - - snd-cs46xx= [HW,ALSA] - - snd-dt019x= [HW,ALSA] - - snd-dummy= [HW,ALSA] - - snd-emu10k1= [HW,ALSA] - - snd-ens1370= [HW,ALSA] - - snd-ens1371= [HW,ALSA] - - snd-es968= [HW,ALSA] - - snd-es1688= [HW,ALSA] - - snd-es18xx= [HW,ALSA] - - snd-es1938= [HW,ALSA] - - snd-es1968= [HW,ALSA] - - snd-fm801= [HW,ALSA] - - snd-gusclassic= [HW,ALSA] - - snd-gusextreme= [HW,ALSA] - - snd-gusmax= [HW,ALSA] - - snd-hdsp= [HW,ALSA] - - snd-ice1712= [HW,ALSA] - - snd-intel8x0= [HW,ALSA] - - snd-interwave= [HW,ALSA] - - snd-interwave-stb= - [HW,ALSA] - - snd-korg1212= [HW,ALSA] - - snd-maestro3= [HW,ALSA] - - snd-mpu401= [HW,ALSA] - - snd-mtpav= [HW,ALSA] - - snd-nm256= [HW,ALSA] - - snd-opl3sa2= [HW,ALSA] - - snd-opti92x-ad1848= - [HW,ALSA] - - snd-opti92x-cs4231= - [HW,ALSA] - - snd-opti93x= [HW,ALSA] - - snd-pmac= [HW,ALSA] - - snd-rme32= [HW,ALSA] - - snd-rme96= [HW,ALSA] - - snd-rme9652= [HW,ALSA] - - snd-sb8= [HW,ALSA] - - snd-sb16= [HW,ALSA] - - snd-sbawe= [HW,ALSA] - - snd-serial= [HW,ALSA] - - snd-sgalaxy= [HW,ALSA] - - snd-sonicvibes= [HW,ALSA] - - snd-sun-amd7930= - [HW,ALSA] - - snd-sun-cs4231= [HW,ALSA] - - snd-trident= [HW,ALSA] - - snd-usb-audio= [HW,ALSA,USB] - - snd-via82xx= [HW,ALSA] - - snd-virmidi= [HW,ALSA] - - snd-wavefront= [HW,ALSA] - - snd-ymfpci= [HW,ALSA] diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html deleted file mode 100644 index 9663b45f6fde..000000000000 --- a/Documentation/sound/alsa/seq_oss.html +++ /dev/null @@ -1,409 +0,0 @@ -<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> -<HTML> -<HEAD> - <TITLE>OSS Sequencer Emulation on ALSA</TITLE> -</HEAD> -<BODY> - -<CENTER> -<H1> - -<HR WIDTH="100%"></H1></CENTER> - -<CENTER> -<H1> -OSS Sequencer Emulation on ALSA</H1></CENTER> - -<HR WIDTH="100%"> -<P>Copyright (c) 1998,1999 by Takashi Iwai -<TT><A HREF="mailto:iwai@ww.uni-erlangen.de"><iwai@ww.uni-erlangen.de></A></TT> -<P>ver.0.1.8; Nov. 16, 1999 -<H2> - -<HR WIDTH="100%"></H2> - -<H2> -1. Description</H2> -This directory contains the OSS sequencer emulation driver on ALSA. Note -that this program is still in the development state. -<P>What this does - it provides the emulation of the OSS sequencer, access -via -<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices. -The most of applications using OSS can run if the appropriate ALSA -sequencer is prepared. -<P>The following features are emulated by this driver: -<UL> -<LI> -Normal sequencer and MIDI events:</LI> - -<BR>They are converted to the ALSA sequencer events, and sent to the corresponding -port. -<LI> -Timer events:</LI> - -<BR>The timer is not selectable by ioctl. The control rate is fixed to -100 regardless of HZ. That is, even on Alpha system, a tick is always -1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>. - -<LI> -Patch loading:</LI> - -<BR>It purely depends on the synth drivers whether it's supported since -the patch loading is realized by callback to the synth driver. -<LI> -I/O controls:</LI> - -<BR>Most of controls are accepted. Some controls -are dependent on the synth driver, as well as even on original OSS.</UL> -Furthermore, you can find the following advanced features: -<UL> -<LI> -Better queue mechanism:</LI> - -<BR>The events are queued before processing them. -<LI> -Multiple applications:</LI> - -<BR>You can run two or more applications simultaneously (even for OSS sequencer)! -However, each MIDI device is exclusive - that is, if a MIDI device is opened -once by some application, other applications can't use it. No such a restriction -in synth devices. -<LI> -Real-time event processing:</LI> - -<BR>The events can be processed in real time without using out of bound -ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed -events will be processed in real-time without queued. To switch off the -real-time mode, send RELTIME 0 event. -<LI> -<TT>/proc</TT> interface:</LI> - -<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT> -at any time. In the later version, configuration will be changed via <TT>/proc</TT> -interface, too.</UL> - -<H2> -2. Installation</H2> -Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>) -and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT> -will be created. If the synth module of your sound card supports for OSS -emulation (so far, only Emu8000 driver), this module will be loaded automatically. -Otherwise, you need to load this module manually. -<P>At beginning, this module probes all the MIDI ports which have been -already connected to the sequencer. Once after that, the creation and deletion -of ports are watched by announcement mechanism of ALSA sequencer. -<P>The available synth and MIDI devices can be found in proc interface. -Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example, -if you use an AWE64 card, you'll see like the following: -<PRE> OSS sequencer emulation version 0.1.8 - ALSA client number 63 - ALSA receiver port 0 - - Number of applications: 0 - - Number of synth devices: 1 - - synth 0: [EMU8000] - type 0x1 : subtype 0x20 : voices 32 - capabilties : ioctl enabled / load_patch enabled - - Number of MIDI devices: 3 - - midi 0: [Emu8000 Port-0] ALSA port 65:0 - capability write / opened none - - midi 1: [Emu8000 Port-1] ALSA port 65:1 - capability write / opened none - - midi 2: [0: MPU-401 (UART)] ALSA port 64:0 - capability read/write / opened none</PRE> -Note that the device number may be different from the information of -<TT>/proc/asound/oss-devices</TT> -or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT> -to play via OSS sequencer emulation. -<H2> -3. Using Synthesizer Devices</H2> -Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 -and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload, -too. -<P>If the lowlevel driver supports multiple access to synth devices (like -Emu8000 driver), two or more applications are allowed to run at the same -time. -<H2> -4. Using MIDI Devices</H2> -So far, only MIDI output was tested. MIDI input was not checked at all, -but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>. -Be aware that these numbers are mostly different from the list in -<TT>/proc/asound/oss-devices</TT>. -<H2> -5. Module Options</H2> -The following module options are available: -<UL> -<LI> -<TT>maxqlen</TT></LI> - -<BR>specifies the maximum read/write queue length. This queue is private -for OSS sequencer, so that it is independent from the queue length of ALSA -sequencer. Default value is 1024. -<LI> -<TT>seq_oss_debug</TT></LI> - -<BR>specifies the debug level and accepts zero (= no debug message) or -positive integer. Default value is 0.</UL> - -<H2> -6. Queue Mechanism</H2> -OSS sequencer emulation uses an ALSA priority queue. The -events from <TT>/dev/sequencer</TT> are processed and put onto the queue -specified by module option. -<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning. -The timing events are also parsed at this moment, so that the events may -be processed in real-time. Sending an event ABSTIME 0 switches the operation -mode to real-time mode, and sending an event RELTIME 0 switches it off. -In the real-time mode, all events are dispatched immediately. -<P>The queued events are dispatched to the corresponding ALSA sequencer -ports after scheduled time by ALSA sequencer dispatcher. -<P>If the write-queue is full, the application sleeps until a certain amount -(as default one half) becomes empty in blocking mode. The synchronization -to write timing was implemented, too. -<P>The input from MIDI devices or echo-back events are stored on read FIFO -queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the -process will be awaked. - -<H2> -7. Interface to Synthesizer Device</H2> - -<H3> -7.1. Registration</H3> -To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT> -function. -<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, - snd_seq_oss_callback_t *oper, void *private_data)</PRE> -The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and -<TT>nvoices</TT> -are used for making the appropriate synth_info structure for ioctl. The -return value is an index number of this device. This index must be remembered -for unregister. If registration is failed, -errno will be returned. -<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>: -<PRE>int snd_seq_oss_synth_unregister(int index),</PRE> -where the <TT>index</TT> is the index number returned by register function. -<H3> -7.2. Callbacks</H3> -OSS synthesizer devices have capability for sample downloading and ioctls -like sample reset. In OSS emulation, these special features are realized -by using callbacks. The registration argument oper is used to specify these -callbacks. The following callback functions must be defined: -<PRE>snd_seq_oss_callback_t: - int (*open)(snd_seq_oss_arg_t *p, void *closure); - int (*close)(snd_seq_oss_arg_t *p); - int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); - int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); - int (*reset)(snd_seq_oss_arg_t *p); -Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed -to be NULL. -<P>Each callback function takes the argument type snd_seq_oss_arg_t as the -first argument. -<PRE>struct snd_seq_oss_arg_t { - int app_index; - int file_mode; - int seq_mode; - snd_seq_addr_t addr; - void *private_data; - int event_passing; -};</PRE> -The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and -<TT>seq_mode</TT> -are initialized by OSS sequencer. The <TT>app_index</TT> is the application -index which is unique to each application opening OSS sequencer. The -<TT>file_mode</TT> -is bit-flags indicating the file operation mode. See -<TT>seq_oss.h</TT> -for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In -the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used. -<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be -filled by the synth driver at open callback. The <TT>addr</TT> contains -the address of ALSA sequencer port which is assigned to this device. If -the driver allocates memory for <TT>private_data</TT>, it must be released -in close callback by itself. -<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on -/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded -as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT> -mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT> -mode checks the note above 128 and regards it as key pressure event (mainly -for Emu8000 driver). -<H4> -7.2.1. Open Callback</H4> -The <TT>open</TT> is called at each time this device is opened by an application -using OSS sequencer. This must not be NULL. Typically, the open callback -does the following procedure: -<OL> -<LI> -Allocate private data record.</LI> - -<LI> -Create an ALSA sequencer port.</LI> - -<LI> -Set the new port address on arg->addr.</LI> - -<LI> -Set the private data record pointer on arg->private_data.</LI> -</OL> -Note that the type bit-flags in port_info of this synth port must NOT contain -<TT>TYPE_MIDI_GENERIC</TT> -bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT> -bit should NOT be included, too. This is necessary to tell it from other -normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, -return -errno. -<H4> -7.2.2 Ioctl Callback</H4> -The <TT>ioctl</TT> callback is called when the sequencer receives device-specific -ioctls. The following two ioctls should be processed by this callback: -<UL> -<LI> -<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI> - -<BR>reset all samples on memory -- return 0 -<LI> -<TT>IOCTL_SYNTH_MEMAVL</TT></LI> - -<BR>return the available memory size -<LI> -<TT>FM_4OP_ENABLE</TT></LI> - -<BR>can be ignored usually</UL> -The other ioctls are processed inside the sequencer without passing to -the lowlevel driver. -<H4> -7.2.3 Load_Patch Callback</H4> -The <TT>load_patch</TT> callback is used for sample-downloading. This callback -must read the data on user-space and transfer to each device. Return 0 -if succeeded, and -errno if failed. The format argument is the patch key -in patch_info record. The buf is user-space pointer where patch_info record -is stored. The offs can be ignored. The count is total data size of this -sample data. -<H4> -7.2.4 Close Callback</H4> -The <TT>close</TT> callback is called when this device is closed by the -application. If any private data was allocated in open callback, it must -be released in the close callback. The deletion of ALSA port should be -done here, too. This callback must not be NULL. -<H4> -7.2.5 Reset Callback</H4> -The <TT>reset</TT> callback is called when sequencer device is reset or -closed by applications. The callback should turn off the sounds on the -relevant port immediately, and initialize the status of the port. If this -callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the -port. -<H3> -7.3 Events</H3> -Most of the events are processed by sequencer and translated to the adequate -ALSA sequencer events, so that each synth device can receive by input_event -callback of ALSA sequencer port. The following ALSA events should be implemented -by the driver: -<BR> -<TABLE BORDER WIDTH="75%" NOSAVE > -<TR NOSAVE> -<TD NOSAVE><B>ALSA event</B></TD> - -<TD><B>Original OSS events</B></TD> -</TR> - -<TR> -<TD>NOTEON</TD> - -<TD>SEQ_NOTEON -<BR>MIDI_NOTEON</TD> -</TR> - -<TR> -<TD>NOTE</TD> - -<TD>SEQ_NOTEOFF -<BR>MIDI_NOTEOFF</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>KEYPRESS</TD> - -<TD>MIDI_KEY_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD>CHANPRESS</TD> - -<TD NOSAVE>SEQ_AFTERTOUCH -<BR>MIDI_CHN_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>PGMCHANGE</TD> - -<TD NOSAVE>SEQ_PGMCHANGE -<BR>MIDI_PGM_CHANGE</TD> -</TR> - -<TR> -<TD>PITCHBEND</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER) -<BR>MIDI_PITCH_BEND</TD> -</TR> - -<TR> -<TD>CONTROLLER</TD> - -<TD>MIDI_CTL_CHANGE -<BR>SEQ_BALANCE (with CTL_PAN)</TD> -</TR> - -<TR> -<TD>CONTROL14</TD> - -<TD>SEQ_CONTROLLER</TD> -</TR> - -<TR> -<TD>REGPARAM</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD> -</TR> - -<TR> -<TD>SYSEX</TD> - -<TD>SEQ_SYSEX</TD> -</TR> -</TABLE> - -<P>The most of these behavior can be realized by MIDI emulation driver -included in the Emu8000 lowlevel driver. In the future release, this module -will be independent. -<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event -type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte -packets without any modification. The lowlevel driver should process these -events appropriately. -<H2> -8. Interface to MIDI Device</H2> -Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer -ports automatically by receiving announcement from ALSA sequencer, the -MIDI devices don't need to be registered explicitly like synth devices. -However, the MIDI port_info registered to ALSA sequencer must include a group -name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or -<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>, -must be defined, too. If these conditions are not satisfied, the port is not -registered as OSS sequencer MIDI device. -<P>The events via MIDI devices are parsed in OSS sequencer and converted -to the corresponding ALSA sequencer events. The input from MIDI sequencer -is also converted to MIDI byte events by OSS sequencer. This works just -a reverse way of seq_midi module. -<H2> -9. Known Problems / TODO's</H2> - -<UL> -<LI> -Patch loading via ALSA instrument layer is not implemented yet.</LI> -</UL> - -</BODY> -</HTML> diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/cards/audigy-mixer.rst index 7f10dc6ff28c..86213234435f 100644 --- a/Documentation/sound/alsa/Audigy-mixer.txt +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -1,8 +1,8 @@ +============================================= +Sound Blaster Audigy mixer / default DSP code +============================================= - Sound Blaster Audigy mixer / default DSP code - =========================================== - -This is based on SB-Live-mixer.txt. +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. @@ -13,8 +13,8 @@ The ALSA driver programs this portion of chip by default code (can be altered later) which offers the following functionality: -1) Digital mixer controls -------------------------- +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 @@ -26,320 +26,343 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped 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. +DAC + digital to analog converter +ADC + analog to digital converter +I2S + one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE + low frequency effects (subwoofer signal) +AC97 + a chip containing an analog mixer, DAC and ADC converters +IEC958 + S/PDIF +FX-bus + the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. name='PCM Front Playback Volume',index=0 - +---------------------------------------- This control is used to attenuate samples for left and right front PCM FX-bus accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM samples for 5.1 playback. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='PCM Surround Playback Volume',index=0 - +------------------------------------------- This control is used to attenuate samples for left and right surround PCM FX-bus accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM slots of the Philips DAC. name='PCM Center Playback Volume',index=0 - +----------------------------------------- This control is used to attenuate samples for center PCM FX-bus accumulator. ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample is forwarded to the center DAC PCM slot of the Philips DAC. name='PCM LFE Playback Volume',index=0 - +-------------------------------------- This control is used to attenuate sample for LFE PCM FX-bus accumulator. ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample is forwarded to the LFE DAC PCM slot of the Philips DAC. name='PCM Playback Volume',index=0 - +---------------------------------- This control is used to attenuate samples for left and right PCM FX-bus accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for stereo playback. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='PCM Capture Volume',index=0 - +--------------------------------- This control is used to attenuate samples for left and right PCM FX-bus accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Music Playback Volume',index=0 - +------------------------------------ This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec. name='Music Capture Volume',index=0 - +----------------------------------- These controls are used to attenuate samples for left and right MIDI FX-bus accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Mic Playback Volume',index=0 - +---------------------------------- This control is used to attenuate samples for left and right Mic input. For Mic input is used AC97 codec. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. name='Mic Capture Volume',index=0 - +--------------------------------- This control is used to attenuate samples for left and right Mic input. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Audigy CD Playback Volume',index=0 - +---------------------------------------- This control is used to attenuate samples from left and right IEC958 TTL digital inputs (usually used by a CDROM drive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='Audigy CD Capture Volume',index=0 - +--------------------------------------- This control is used to attenuate samples from left and right IEC958 TTL digital inputs (usually used by a CDROM drive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='IEC958 Optical Playback Volume',index=0 - +--------------------------------------------- This control is used to attenuate samples from left and right IEC958 optical digital input. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='IEC958 Optical Capture Volume',index=0 - +-------------------------------------------- This control is used to attenuate samples from left and right IEC958 optical digital inputs. The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Line2 Playback Volume',index=0 - +------------------------------------ This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='Line2 Capture Volume',index=1 - +----------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Analog Mix Playback Volume',index=0 - +----------------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs from Philips ADC. The result samples are forwarded to the front DAC PCM slots of the Philips DAC. This contains mix from analog sources like CD, Line In, Aux, .... name='Analog Mix Capture Volume',index=1 - +---------------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs Philips ADC. The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Aux2 Playback Volume',index=0 - +----------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the front DAC PCM slots of the Philips DAC. name='Aux2 Capture Volume',index=1 - +---------------------------------- This control is used to attenuate samples from left and right I2S ADC inputs (on the AudigyDrive). The result samples are forwarded to the ADC capture FIFO (thus to the standard capture PCM device). name='Front Playback Volume',index=0 - +------------------------------------ All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate samples for left and right front speakers of this mix. name='Surround Playback Volume',index=0 - +--------------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate samples for left and right surround speakers of this mix. name='Center Playback Volume',index=0 - +------------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate sample for center speaker of this mix. name='LFE Playback Volume',index=0 - +---------------------------------- All stereo signals are mixed together and mirrored to surround, center and LFE. This control is used to attenuate sample for LFE speaker of this mix. name='Tone Control - Switch',index=0 - +------------------------------------ This control turns the tone control on or off. The samples for front, rear and center / LFE outputs are affected. name='Tone Control - Bass',index=0 - +---------------------------------- This control sets the bass intensity. There is no neutral value!! When the tone control code is activated, the samples are always modified. The closest value to pure signal is 20. name='Tone Control - Treble',index=0 - +------------------------------------ This control sets the treble intensity. There is no neutral value!! When the tone control code is activated, the samples are always modified. The closest value to pure signal is 20. name='Master Playback Volume',index=0 - +------------------------------------- This control is used to attenuate samples for front, surround, center and LFE outputs. name='IEC958 Optical Raw Playback Switch',index=0 - +------------------------------------------------- If this switch is on, then the samples for the IEC958 (S/PDIF) digital output are taken only from the raw FX8010 PCM, otherwise standard front PCM samples are taken. -2) PCM stream related controls ------------------------------- +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) +* 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 +* 0 - mono, A destination (FX-bus 0-63), default 0 +* 1 - mono, B destination (FX-bus 0-63), default 1 +* 2 - mono, C destination (FX-bus 0-63), default 2 +* 3 - mono, D destination (FX-bus 0-63), default 3 +* 4 - mono, E destination (FX-bus 0-63), default 0 +* 5 - mono, F destination (FX-bus 0-63), default 0 +* 6 - mono, G destination (FX-bus 0-63), default 0 +* 7 - mono, H destination (FX-bus 0-63), default 0 +* 8 - left, A destination (FX-bus 0-63), default 0 +* 9 - left, B destination (FX-bus 0-63), default 1 +* 10 - left, C destination (FX-bus 0-63), default 2 +* 11 - left, D destination (FX-bus 0-63), default 3 +* 12 - left, E destination (FX-bus 0-63), default 0 +* 13 - left, F destination (FX-bus 0-63), default 0 +* 14 - left, G destination (FX-bus 0-63), default 0 +* 15 - left, H destination (FX-bus 0-63), default 0 +* 16 - right, A destination (FX-bus 0-63), default 0 +* 17 - right, B destination (FX-bus 0-63), default 1 +* 18 - right, C destination (FX-bus 0-63), default 2 +* 19 - right, D destination (FX-bus 0-63), default 3 +* 20 - right, E destination (FX-bus 0-63), default 0 +* 21 - right, F destination (FX-bus 0-63), default 0 +* 22 - right, G destination (FX-bus 0-63), default 0 +* 23 - right, H destination (FX-bus 0-63), default 0 Don't forget that it's illegal to assign a channel to the same FX-bus accumulator more than once (it means 0=0 && 1=0 is an invalid combination). name='EMU10K1 PCM Send Volume',index 0-31 - +----------------------------------------- It specifies the attenuation (amount) for given destination in range 0-255. The channel mapping is following: - 0 - mono, A destination attn, default 255 (no attenuation) - 1 - mono, B destination attn, default 255 (no attenuation) - 2 - mono, C destination attn, default 0 (mute) - 3 - mono, D destination attn, default 0 (mute) - 4 - mono, E destination attn, default 0 (mute) - 5 - mono, F destination attn, default 0 (mute) - 6 - mono, G destination attn, default 0 (mute) - 7 - mono, H destination attn, default 0 (mute) - 8 - left, A destination attn, default 255 (no attenuation) - 9 - left, B destination attn, default 0 (mute) - 10 - left, C destination attn, default 0 (mute) - 11 - left, D destination attn, default 0 (mute) - 12 - left, E destination attn, default 0 (mute) - 13 - left, F destination attn, default 0 (mute) - 14 - left, G destination attn, default 0 (mute) - 15 - left, H destination attn, default 0 (mute) - 16 - right, A destination attn, default 0 (mute) - 17 - right, B destination attn, default 255 (no attenuation) - 18 - right, C destination attn, default 0 (mute) - 19 - right, D destination attn, default 0 (mute) - 20 - right, E destination attn, default 0 (mute) - 21 - right, F destination attn, default 0 (mute) - 22 - right, G destination attn, default 0 (mute) - 23 - right, H destination attn, default 0 (mute) - - - -4) MANUALS/PATENTS: -------------------- +* 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 ------------------------------------- - Files: - LM4545.pdf AC97 Codec +LM4545.pdf + AC97 Codec - m2049.pdf The EMU10K1 Digital Audio Processor +m2049.pdf + The EMU10K1 Digital Audio Processor - hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects WIPO Patents ------------ - Patent numbers: - WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) - streams - WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) +WO 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) +WO 9901953 (A1) + Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) US Patents (http://www.uspto.gov/) ---------------------------------- - US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) - - US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) - with a multiport memory onto which multiple asynchronous - digital sound samples can be concurrently loaded - - US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) - - US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) - - US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) - system bus with prioritization and modification of bus transfers - in accordance with loop ends and minimum block sizes - - US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) - pool of short term memory registers - - US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) - a common interrupt by associating programs to GP registers, - defining interrupt register, polling GP registers, and invoking - callback routine associated with defined interrupt register +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/alsa/Audiophile-Usb.txt b/Documentation/sound/cards/audiophile-usb.rst index e7a5ed4dcae8..a7bb5648331f 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/cards/audiophile-usb.rst @@ -1,32 +1,41 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 - ======================================================== +======================================================== +Guide to using M-Audio Audiophile USB with ALSA and Jack +======================================================== - Thibault Le Meur <Thibault.LeMeur@supelec.fr> +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 + + - 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 + - Added AC3/DTS passthru info -1 - Audiophile USB Specs and correct usage -========================================== +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) @@ -34,56 +43,69 @@ The device has 4 audio interfaces, and 2 MIDI ports: * 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" + 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 + 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 + 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 + 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 + 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) + synchronization error (for instance sound played at an odd sample rate) -2 - Audiophile USB MIDI support in ALSA -======================================= +Audiophile USB MIDI support in ALSA +=================================== The Audiophile USB MIDI ports will be automatically supported once the following modules have been loaded: + * snd-usb-audio * snd-seq-midi No additional setting is required. -3 - Audiophile USB Audio support in ALSA -======================================== +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". +``device_setup``. -3.1 - Default Alsa driver mode ------------------------------- +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 @@ -101,6 +123,7 @@ 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 @@ -115,20 +138,28 @@ 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 + + * 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 + + * 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 + + * 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 + + * playing an ac3 sample file to the Do port:: + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw If you're happy with the default Alsa driver mode and don't experience any issue with this mode, then you can skip the following chapter. -3.2 - Advanced module setup ---------------------------- +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 @@ -137,34 +168,39 @@ 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 +parameter called ``device_setup`` (this parameter was introduced in kernel release 2.6.17) -3.2.1 - Initializing the working mode of the Audiophile USB +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 +When initialized with ``device_setup=0x00``, the snd-usb-audio module has the same behaviour as when the parameter is omitted (see paragraph "Default Alsa driver mode" above) Others modes are described in the following subsections. -3.2.1.1 - 16-bit modes +16-bit modes +~~~~~~~~~~~~ The two supported modes are: - * device_setup=0x01 + * ``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 + * ``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 @@ -173,33 +209,43 @@ The two supported modes are: In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, the devices where reported to be Big-Endian when in fact they were Little-Endian so that playing a file was a matter of using: +:: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw + where "test_S16_LE.raw" was in fact a little-endian sample file. Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in these modes) a fix has been committed (expected in kernel 2.6.23) and Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as using: +:: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw -3.2.1.2 - 24-bit modes + +24-bit modes +~~~~~~~~~~~~ The three supported modes are: - * device_setup=0x09 + * ``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 + * ``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 + * ``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 @@ -210,29 +256,35 @@ The three supported modes are: In these modes the device is only Big-Endian compliant (see "Default Alsa driver mode" above for an aplay command example) -3.2.1.3 - AC3 w/ DTS passthru mode +AC3 w/ DTS passthru mode +~~~~~~~~~~~~~~~~~~~~~~~~ Thanks to Hakan Lennestal, I now have a report saying that this mode works. - * device_setup=0x03 + * ``device_setup=0x03`` + - 16bits 48kHz mode with only the Do port enabled - AC3 with DTS passthru - Caution with this setup the Do port is mapped to the pcm device hw:1,0 The command line used to playback the AC3/DTS encoded .wav-files in this mode: +:: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw -3.2.2 - How to use the device_setup parameter ----------------------------------------------- +How to use the ``device_setup`` parameter +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ The parameter can be given: - * By manually probing the device (as root): + * 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: + (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 @@ -250,26 +302,31 @@ CAUTION when initializing the device * 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 + option in ``/etc/modprobe.d/*.conf`` - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not be enough to ensure the 'stability' of the device initialization. -3.2.3 - Technical details for hackers -------------------------------------- +Technical details for hackers +----------------------------- + This section is for hackers, wanting to understand details about the device internals and how Alsa supports it. -3.2.3.1 - Audiophile USB's device_setup structure +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| @@ -278,38 +335,55 @@ The device_setup is one byte long and its structure is the following: +---+---+---+---+---+---+---+---+ Where: - * b0 is the "SET" bit + + * b0 is the ``SET`` bit + - it MUST be set if device_setup is initialized - * b1 is the "DTS" bit + + * 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 + + - 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 + + - 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 + + - 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" + + * b5 to b7 are reserved for future uses, and must be set to ``0`` + - might become Ao, Do, Ai, for b7, b6, b4 respectively Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll only be able to use one at the same time -3.2.3.2 - USB implementation details for this device +USB implementation details for this device +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ You may safely skip this section if you're not interested in driver hacking. @@ -319,46 +393,72 @@ 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 + +* 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 + +* AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode - Asynch capture and playback (Ao,Ai,Do,Di) - * AltSettings 3 corresponds to + +* AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode - Synch capture (Ai) and Adaptive playback (Ao,Do) - * AltSettings 4 corresponds to + +* AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode - Asynch capture and playback (Ao,Ai,Do,Di) - * AltSettings 5 corresponds to + +* AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode - Synch capture (Ai) and Adaptive playback (Ao,Do) - * AltSettings 6 corresponds to + +* 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: +*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 @@ -371,20 +471,21 @@ _must_know_ how the device will be used: 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 +``audiophile_skip_setting_quirk`` in order to prevent AltSettings not corresponding to device_setup from being registered in the driver. -4 - Audiophile USB and Jack support -=================================== +Audiophile USB and Jack support +=============================== This section deals with support of the Audiophile USB device in Jack. There are 2 main potential issues when using Jackd with the device: + * support for Big-Endian devices in 24-bit modes * support for 4-in / 4-out channels -4.1 - Direct support in Jackd ------------------------------ +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 @@ -396,29 +497,35 @@ are now Little Endians ;-) ). You can run jackd with the following command for playback with Ao and record with Ai: +:: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 -4.2 - Using Alsa plughw ------------------------ +Using Alsa plughw +----------------- + If you don't have a recent Jackd installed, you can downgrade to using -the Alsa "plug" converter. +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." + You appear to be using the ALSA software "plug" layer, probably a result of + using the "default" ALSA device. This is less efficient than it could be. + Consider using a hardware device instead rather than using the plug layer. -4.3 - Getting 2 input and/or output interfaces in Jack ------------------------------------------------------- +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) @@ -432,6 +539,7 @@ 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 diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/cards/bt87x.rst index f158cde8b065..912732d3ef9e 100644 --- a/Documentation/sound/alsa/Bt87x.txt +++ b/Documentation/sound/cards/bt87x.rst @@ -1,18 +1,23 @@ +================= +ALSA BT87x Driver +================= + Intro ===== You might have noticed that the bt878 grabber cards have actually -_two_ PCI functions: +*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) -[ ... ] + $ 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 +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... @@ -30,9 +35,9 @@ 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 +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 +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 diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/cards/cmipci.rst index 4e36e6e809ca..9ea1de6ec4ce 100644 --- a/Documentation/sound/alsa/CMIPCI.txt +++ b/Documentation/sound/cards/cmipci.rst @@ -1,7 +1,8 @@ - Brief Notes on C-Media 8338/8738/8768/8770 Driver - ================================================= +================================================= +Brief Notes on C-Media 8338/8738/8768/8770 Driver +================================================= - Takashi Iwai <tiwai@suse.de> +Takashi Iwai <tiwai@suse.de> Front/Rear Multi-channel Playback @@ -30,19 +31,20 @@ The rear output can be heard only when "Four Channel Mode" switch is disabled. Otherwise no signal will be routed to the rear speakers. As default it's turned on. -*** WARNING *** -When "Four Channel Mode" switch is off, the output from rear speakers -will be FULL VOLUME regardless of Master and PCM volumes. -This might damage your audio equipment. Please disconnect speakers -before your turn off this switch. -*** WARNING *** +.. 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 + +.. [#] + 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... ] + 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 @@ -73,12 +75,14 @@ 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 @@ -89,13 +93,15 @@ 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 +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 +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 @@ -111,11 +117,14 @@ 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. @@ -140,31 +149,40 @@ 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 +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) +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 +IEC958 Loop + SPDIF input data is loop back into SPDIF output (aka bypass) -"IEC958 Copyright" - Set the copyright bit. +IEC958 Copyright + Set the copyright bit. -"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface. +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 Monitor + SPDIF input is routed to DAC. -"IEC958 In Phase Inverse" - Set SPDIF input format as inverse. +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. +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. @@ -217,7 +235,7 @@ to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and 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 +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 diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/cards/emu10k1-jack.rst index 751d45036a05..6597f1ea83f0 100644 --- a/Documentation/sound/alsa/emu10k1-jack.txt +++ b/Documentation/sound/cards/emu10k1-jack.rst @@ -1,3 +1,7 @@ +================================================================= +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 @@ -7,8 +11,6 @@ power of this hardware. http://www.kxproject.com - Lee Revell, 2005.03.30 -Low latency, multichannel audio with JACK and the emu10k1/emu10k2 ------------------------------------------------------------------ Until recently, emu10k1 users on Linux did not have access to the same low latency, multichannel features offered by the "kX ASIO" feature of their @@ -23,14 +25,15 @@ 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 + /usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S This will give you 16 input ports and 16 output ports. The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the Audigy). The mapping from FX bus to physical output is described in -SB-Live-mixer.txt (or Audigy-mixer.txt). +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 @@ -49,10 +52,11 @@ 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 (& 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) @@ -69,6 +73,6 @@ capture_13 asio10 FXBUS2(0xa) capture_14 asio11 FXBUS2(0xb) capture_15 asio12 FXBUS2(0xc) capture_16 asio13 FXBUS2(0xd) -*/ +============== ======== ============ TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/cards/hdspm.rst index 7ba31948dea7..5373e51ed076 100644 --- a/Documentation/sound/alsa/hdspm.txt +++ b/Documentation/sound/cards/hdspm.rst @@ -1,21 +1,24 @@ +======================================= Software Interface ALSA-DSP MADI Driver +======================================= (translated from German, so no good English ;-), -2004 - winfried ritsch - +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. +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: +Hardware functionality +====================== - Audio transmission: +Audio transmission +------------------ - number of channels -- depends on transmission mode +* 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, @@ -23,31 +26,34 @@ Software Interface ALSA-DSP MADI Driver allocated. So also the throughput of the PCI system can be scaled. (Only important for low performance boards). - Single Speed -- 1..64 channels +* 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 +* 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 +* Quad Speed -- 1..16 channels - Note: Choosing the 56-channel mode for +.. 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) +* Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) - Sample Rates -- +* Sample Rates -- Single Speed -- 32000, 44100, 48000 @@ -55,14 +61,13 @@ Software Interface ALSA-DSP MADI Driver Quad Speed -- 128000, 176400, 192000 (untested) - access-mode -- MMAP (memory mapped), Not interleaved - (PCM_NON-INTERLEAVED) +* access-mode -- MMAP (memory mapped), Not interleaved (PCM_NON-INTERLEAVED) - buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples +* buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples - fragments -- 2 +* fragments -- 2 - Hardware-pointer -- 2 Modi +* Hardware-pointer -- 2 Modi The Card supports the readout of the actual Buffer-pointer, @@ -74,53 +79,54 @@ Software Interface ALSA-DSP MADI Driver 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 + * Precise Pointer -- off interrupt used for pointer-calculation - - Precise Pointer -- on + + * Precise Pointer -- on hardware pointer used. - Controller: - +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. +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 !!! +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" +* System Clock -- suspended !!!! - Access -- Read Write - - Values -- "Master" "Slave" + * 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 @@ -128,106 +134,107 @@ Software Interface ALSA-DSP MADI Driver a studio should have working synchronisations setup. So use Clock-source-controller instead !!!! - Clock Source +* Clock Source - Name -- "Sample Clock Source" + * Name -- "Sample Clock Source" - Access -- Read Write + * 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" + * 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 +* Preferred Sync Ref - Name -- "Preferred Sync Reference" + * Name -- "Preferred Sync Reference" - Access -- Read Write + * Access -- Read Write - Values -- "Word" "MADI" + * 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 --- +* TX 64 channel - Name -- "TX 64 channels mode" + * Name -- "TX 64 channels mode" - Access -- Read Write + * Access -- Read Write - Values -- 0 1 + * 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 --- +* Clear TMS - Name -- "Clear Track Marker" + * Name -- "Clear Track Marker" - Access -- Read Write + * Access -- Read Write - Values -- 0 1 + * Values -- 0 1 Don't use to lower 5 Audio-bits on AES as additional Bits. - Safe Mode oder Auto Input --- +* Safe Mode oder Auto Input - Name -- "Safe Mode" + * Name -- "Safe Mode" - Access -- Read Write + * Access -- Read Write - Values -- 0 1 - - (default on) + * 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 --- +* Input - Name -- "Input Select" + * Name -- "Input Select" - Access -- Read Write + * Access -- Read Write - Values -- optical coaxial + * Values -- optical coaxial Choosing the Input, optical or coaxial. If Safe-mode is active, this is the preferred Input. --------------- Mixer ---------------------- +Mixer +----- - Mixer +* Mixer - Name -- "Mixer" + * Name -- "Mixer" - Access -- Read Write + * Access -- Read Write - Values - <channel-number 0-127> <Value 0-65535> + * Values - <channel-number 0-127> <Value 0-65535> Here as a first value the channel-index is taken to get/set the @@ -235,40 +242,41 @@ Software Interface ALSA-DSP MADI Driver fader and 64-127 the playback to outputs fader. Value 0 is channel muted 0 and 32768 an amplification of 1. - Chn 1-64 +* Chn 1-64 fast mixer for the ALSA-mixer utils. The diagonal of the mixer-matrix is implemented from playback to output. - Line Out +* Line Out - Name -- "Line Out" + * Name -- "Line Out" - Access -- Read Write + * Access -- Read Write - Values -- 0 1 + * 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): +Information (only read access) +------------------------------ - Sample Rate +* Sample Rate - Name -- "System Sample Rate" + * Name -- "System Sample Rate" - Access -- Read-only + * Access -- Read-only getting the sample rate. - External Rate measured +* External Rate measured - Name -- "External Rate" + * Name -- "External Rate" - Access -- Read only + * Access -- Read only Should be "Autosync Rate", but Name used is @@ -276,79 +284,86 @@ Software Interface ALSA-DSP MADI Driver reported. - MADI Sync Status +* MADI Sync Status - Name -- "MADI Sync Lock Status" + * Name -- "MADI Sync Lock Status" - Access -- Read + * Access -- Read - Values -- 0,1,2 + * Values -- 0,1,2 MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. - Word Clock Sync Status +* Word Clock Sync Status - Name -- "Word Clock Lock Status" + * Name -- "Word Clock Lock Status" - Access -- Read + * Access -- Read - Values -- 0,1,2 + * Values -- 0,1,2 Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. - AutoSync +* AutoSync - Name -- "AutoSync Reference" + * Name -- "AutoSync Reference" - Access -- Read + * Access -- Read - Values -- "WordClock", "MADI", "None" + * Values -- "WordClock", "MADI", "None" Sync-Reference is either "WordClock", "MADI" or none. - RX 64ch --- noch nicht implementiert +* RX 64ch --- noch nicht implementiert MADI-Receiver is in 64 channel mode oder 56 channel mode. - AB_inp --- not tested +* AB_inp --- not tested Used input for Auto-Input. - actual Buffer Position --- not implemented +* actual Buffer Position --- not implemented !!! this is a ALSA internal function, so no control is used !!! -Calling Parameter: +Calling Parameter +================= + +* index int array (min = 1, max = 8) - index int array (min = 1, max = 8), - "Index value for RME HDSPM interface." card-index within ALSA + 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." +* 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." +* 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." +* 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." +* 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 @@ -356,7 +371,9 @@ Calling Parameter: - enable_monitor int array (min = 1, max = 8), - "Enable Analog Out on Channel 63/64 by default." +* 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/alsa/img,spdif-in.txt b/Documentation/sound/cards/img-spdif-in.rst index 8b7505785fa6..7df9f5ae2609 100644 --- a/Documentation/sound/alsa/img,spdif-in.txt +++ b/Documentation/sound/cards/img-spdif-in.rst @@ -1,21 +1,25 @@ +================================================ +Imagination Technologies SPDIF Input Controllers +================================================ + The Imagination Technologies SPDIF Input controller contains the following controls: -name='IEC958 Capture Mask',index=0 +* 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 +* 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 +* 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' @@ -29,21 +33,21 @@ 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 +* 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 +* 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 +* 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 +* 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 000000000000..c016f8c3b88b --- /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/alsa/Joystick.txt b/Documentation/sound/cards/joystick.rst index ccda41b10f8a..a6e468c81d02 100644 --- a/Documentation/sound/alsa/Joystick.txt +++ b/Documentation/sound/cards/joystick.rst @@ -1,7 +1,10 @@ +======================================= Analog Joystick Support on ALSA Drivers ======================================= - Oct. 14, 2003 - Takashi Iwai <tiwai@suse.de> + +Oct. 14, 2003 + +Takashi Iwai <tiwai@suse.de> General ------- @@ -34,44 +37,46 @@ stability and the resource management. The following PCI drivers support the joystick natively. - Driver Module Option Available Values - --------------------------------------------------------------------------- - als4000 joystick_port 0 = disable (default), 1 = auto-detect, - manual: any address (e.g. 0x200) - au88x0 N/A N/A - azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) - ens1370 joystick 0 = disable (default), 1 = enable - ens1371 joystick_port 0 = disable (default), 1 = auto-detect, - manual: 0x200, 0x208, 0x210, 0x218 - cmipci joystick_port 0 = disable (default), 1 = auto-detect, - manual: any address (e.g. 0x200) - cs4281 N/A N/A - cs46xx N/A N/A - es1938 N/A N/A - es1968 joystick 0 = disable (default), 1 = enable - sonicvibes N/A N/A - trident N/A N/A - via82xx(*1) joystick 0 = disable (default), 1 = enable - ymfpci joystick_port 0 = disable (default), 1 = auto-detect, - manual: 0x201, 0x202, 0x204, 0x205(*2) - --------------------------------------------------------------------------- - - *1) VIA686A/B only - *2) With YMF744/754 chips, the port address can be chosen arbitrarily +============== ============= ============================================ +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 - ----------------------------- +======= ================= +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. +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" diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/cards/maya44.rst index 67b2ea1cc31d..bf09a584b443 100644 --- a/Documentation/sound/alsa/README.maya44 +++ b/Documentation/sound/cards/maya44.rst @@ -1,10 +1,18 @@ -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 +================================= +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: +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). @@ -44,16 +52,17 @@ Things that do not seem to work: - 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: +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 +* 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. @@ -62,25 +71,26 @@ This is done in maya44.c, mainly because some of the WM8776 controls are used in 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 +* 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: +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. @@ -98,66 +108,79 @@ I propose some additional code for limiting the sampling rate when setting on a The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). -SOUND DEVICES: +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) +* 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: +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 +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. -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). +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. -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 -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. -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: - +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 + 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. -H/W 4 ... H/W 9: unknown function, left in to enable testing. Possibly some of these control S/PDIF output(s). If these turn out to be unused, they will go away in later driver versions. Selectable values for each of the digital output selectors are: - "PCM out" -> DAC output of the corresponding channel (default setting) - "Input 1"... - "Input 4" -> direct routing from ADC output of the selected input channel - --------- - -Feb 14, 2008 -Rainer Zimmermann -mail@lightshed.de +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/alsa/MIXART.txt b/Documentation/sound/cards/mixart.rst index 4ee35b4fbe4a..48aba98b088f 100644 --- a/Documentation/sound/alsa/MIXART.txt +++ b/Documentation/sound/cards/mixart.rst @@ -1,5 +1,8 @@ - Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards - Digigram <alsa@digigram.com> +============================================================== +Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards +============================================================== + +Digigram <alsa@digigram.com> GENERAL @@ -48,11 +51,15 @@ 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. +<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 @@ -79,11 +86,14 @@ FIRMWARE 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.) +/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 diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/cards/sb-live-mixer.rst index f4b5988f450c..bcb62fc99bbb 100644 --- a/Documentation/sound/alsa/SB-Live-mixer.txt +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -1,6 +1,6 @@ - - Sound Blaster Live mixer / default DSP code - =========================================== +=========================================== +Sound Blaster Live mixer / default DSP code +=========================================== The EMU10K1 chips have a DSP part which can be programmed to support @@ -12,8 +12,8 @@ The ALSA driver programs this portion of chip by default code (can be altered later) which offers the following functionality: -1) IEC958 (S/PDIF) raw PCM --------------------------- +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 @@ -27,8 +27,8 @@ at the time. Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. -2) Digital mixer controls -------------------------- +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 @@ -40,317 +40,334 @@ is mentioned in multiple controls, the signal is accumulated and can be wrapped 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 - +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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 +.. 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 +.. 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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``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 - +``name='Headphone Playback Volume',index=1`` +-------------------------------------------- This control attenuates the samples for the headphone output. -name='Headphone Center Playback Switch',index=1 - +``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 - +``name='Headphone LFE Playback Switch',index=1`` +------------------------------------------------ If this switch is on, then the sample for the center PCM is put to the right headphone output (useful for SB Live cards without separate center/LFE output). -3) PCM stream related controls ------------------------------- - -name='EMU10K1 PCM Volume',index 0-31 +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 +* 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 +* 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 - +``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) +* 0 - mono, A destination attn, default 255 (no attenuation) +* 1 - mono, B destination attn, default 255 (no attenuation) +* 2 - mono, C destination attn, default 0 (mute) +* 3 - mono, D destination attn, default 0 (mute) +* 4 - left, A destination attn, default 255 (no attenuation) +* 5 - left, B destination attn, default 0 (mute) +* 6 - left, C destination attn, default 0 (mute) +* 7 - left, D destination attn, default 0 (mute) +* 8 - right, A destination attn, default 0 (mute) +* 9 - right, B destination attn, default 255 (no attenuation) +* 10 - right, C destination attn, default 0 (mute) +* 11 - right, D destination attn, default 0 (mute) -4) MANUALS/PATENTS: -------------------- +MANUALS/PATENTS +=============== ftp://opensource.creative.com/pub/doc ------------------------------------- - Files: - LM4545.pdf AC97 Codec - - m2049.pdf The EMU10K1 Digital Audio Processor - - hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects +LM4545.pdf + AC97 Codec +m2049.pdf + The EMU10K1 Digital Audio Processor +hog63.ps + FX8010 - A DSP Chip Architecture for Audio Effects WIPO Patents ------------ - Patent numbers: - WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) - streams - WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) +WO 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) +WO 9901953 (A1) + Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) US Patents (http://www.uspto.gov/) ---------------------------------- - US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) - - US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) - with a multiport memory onto which multiple asynchronous - digital sound samples can be concurrently loaded - - US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) - - US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) - - US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) - system bus with prioritization and modification of bus transfers - in accordance with loop ends and minimum block sizes - - US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) - pool of short term memory registers - - US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) - a common interrupt by associating programs to GP registers, - defining interrupt register, polling GP registers, and invoking - callback routine associated with defined interrupt register +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/alsa/serial-u16550.txt b/Documentation/sound/cards/serial-u16550.rst index c1919559d509..197aeacea3da 100644 --- a/Documentation/sound/alsa/serial-u16550.txt +++ b/Documentation/sound/cards/serial-u16550.rst @@ -1,14 +1,14 @@ - - Serial UART 16450/16550 MIDI driver - =================================== +=================================== +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) +* 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. @@ -22,11 +22,13 @@ 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 @@ -37,6 +39,7 @@ 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 \ @@ -47,6 +50,7 @@ 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 @@ -63,6 +67,7 @@ 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 diff --git a/Documentation/sound/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst new file mode 100644 index 000000000000..6ee993d4535b --- /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/alsa/Channel-Mapping-API.txt b/Documentation/sound/designs/channel-mapping-api.rst index 3c43d1a4ca0e..58e6312a43c0 100644 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ b/Documentation/sound/designs/channel-mapping-api.rst @@ -1,9 +1,11 @@ +============================ ALSA PCM channel-mapping API ============================ - Takashi Iwai <tiwai@suse.de> -GENERAL -------- +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 @@ -11,9 +13,9 @@ 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 } +``{ front_left, front_right }`` while a 4.0 surround PCM stream has a channel map of - { front left, front right, rear left, rear right }. +``{ 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 @@ -29,8 +31,8 @@ specification. These are the main motivations for the new channel mapping API. -DESIGN ------- +Design +====== Actually, "the channel mapping API" doesn't introduce anything new in the kernel/user-space ABI perspective. It uses only the existing @@ -39,10 +41,11 @@ 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 + +* 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. @@ -50,32 +53,35 @@ 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 +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... +``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 +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 +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. +These new TLV types are defined in ``sound/tlv.h``. -The available channel position values are defined in sound/asound.h, +The available channel position values are defined in ``sound/asound.h``, here is a cut: -/* channel positions */ -enum { +:: + + /* channel positions */ + enum { SNDRV_CHMAP_UNKNOWN = 0, SNDRV_CHMAP_NA, /* N/A, silent */ SNDRV_CHMAP_MONO, /* mono stream */ @@ -107,11 +113,13 @@ enum { 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 @@ -120,19 +128,21 @@ to be returned will contain such as: 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) + #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, +``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 +When ``SNDRV_CHMAP_DRIVER_SPEC`` is set, all the channel position values don't follow the standard definition above but driver-specific. -* READ OPERATION +Read Operation +-------------- The control read operation is for providing the current channel map of the given stream. The control element returns an integer array @@ -140,9 +150,10 @@ 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. +``UNKNOWN``. -* WRITE OPERATION +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 diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/designs/compress-offload.rst index 8ba556a131c3..ad4bfbdacc83 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/designs/compress-offload.rst @@ -1,10 +1,14 @@ - compress_offload.txt - ===================== - Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com> - Vinod Koul <vinod.koul@linux.intel.com> +========================= +ALSA Compress-Offload API +========================= + +Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com> + +Vinod Koul <vinod.koul@linux.intel.com> -Overview +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 @@ -27,8 +31,9 @@ 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 +Requirements +============ The main requirements are: - separation between byte counts and time. Compressed formats may have @@ -63,7 +68,7 @@ The main requirements are: 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 +- 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 @@ -72,7 +77,7 @@ The main requirements are: 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. @@ -95,43 +100,44 @@ 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. +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. @@ -145,6 +151,7 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) + Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder @@ -162,19 +169,19 @@ 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_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 +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 +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 @@ -189,10 +196,12 @@ Sequence flow for gapless would be: - then call partial_drain to flush most of buffer in DSP - Fill data of the next track - DSP switches to second track + (note: order for partial_drain and write for next track can be reversed as well) -Not supported: +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. @@ -225,7 +234,9 @@ Not supported: rendered output in time, this does not deal with underrun/overrun and maybe dealt in user-library -Credits: + +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 diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst new file mode 100644 index 000000000000..7fedd0f33cd9 --- /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 cannel +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 000000000000..04dcdae3e4f2 --- /dev/null +++ b/Documentation/sound/designs/index.rst @@ -0,0 +1,15 @@ +Designs and Implementations +=========================== + +.. toctree:: + :maxdepth: 2 + + control-names + channel-mapping-api + compress-offload + timestamping + jack-controls + procfile + powersave + oss-emulation + seq-oss diff --git a/Documentation/sound/alsa/Jack-Controls.txt b/Documentation/sound/designs/jack-controls.rst index fe1c5e0c8555..ae25b1531bb0 100644 --- a/Documentation/sound/alsa/Jack-Controls.txt +++ b/Documentation/sound/designs/jack-controls.rst @@ -1,3 +1,7 @@ +================== +ALSA Jack Controls +================== + Why we need Jack kcontrols ========================== @@ -29,11 +33,12 @@ How to use jack kcontrols ========================= In order to keep compatibility, snd_jack_new() has been modified by -adding two params :- +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. +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 diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/designs/oss-emulation.rst index 152ca2a3f1bd..e8dcb9633e7b 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/designs/oss-emulation.rst @@ -1,7 +1,8 @@ - NOTES ON KERNEL OSS-EMULATION - ============================= +============================= +Notes on Kernel OSS-Emulation +============================= - Jan. 22, 2004 Takashi Iwai <tiwai@suse.de> +Jan. 22, 2004 Takashi Iwai <tiwai@suse.de> Modules @@ -14,18 +15,18 @@ 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 +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 +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. +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 +``alias sound-slot-0 snd-card-0`` doesn't work any more like the old modutils). The currently available OSS configuration is shown in @@ -42,6 +43,7 @@ Device Mapping ============== ALSA supports the following OSS device files: +:: PCM: /dev/dspX @@ -61,48 +63,55 @@ ALSA supports the following OSS device files: 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.) +/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 +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) +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, +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) +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 @@ -118,43 +127,52 @@ 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 +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 +``app_name`` is the name of application with (higher priority) or without path. -fragments specifies the number of fragments or zero if no specific +``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 +``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 +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 +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 @@ -171,10 +189,11 @@ 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. +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 @@ -183,15 +202,18 @@ 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 +Also, for clearing all the current configuration, send ``erase`` command as below: +:: % echo "erase" > /proc/asound/card0/pcm0p/oss @@ -211,40 +233,43 @@ 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 +==================== ===================== ===== +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. +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 @@ -261,6 +286,7 @@ 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 @@ -284,12 +310,18 @@ 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. @@ -302,4 +334,3 @@ ICE1712 supports only the unconventional format, interleaved 10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap the buffer as the conventional (mono or 2-channels, 8 or 16bit) format on OSS. - diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/designs/powersave.rst index 9657e8099228..138157452eb9 100644 --- a/Documentation/sound/alsa/powersave.txt +++ b/Documentation/sound/designs/powersave.rst @@ -1,9 +1,10 @@ +========================== 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. +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 @@ -11,20 +12,21 @@ 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 +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 +``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 +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): +``/sys/modules/snd_ac97_codec/parameters/power_save`` (usually as root): +:: # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/designs/procfile.rst index 7f8a0d325905..29a466851fd2 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/designs/procfile.rst @@ -1,20 +1,22 @@ - Proc Files of ALSA Drivers - ========================== - Takashi Iwai <tiwai@suse.de> +========================== +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. +card-specific files are stored in the ``card*`` subdirectories. Global Information ------------------- +================== cards Shows the list of currently configured ALSA drivers, @@ -31,15 +33,15 @@ devices meminfo Shows the status of allocated pages via ALSA drivers. - Appears only when CONFIG_SND_DEBUG=y. + Appears only when ``CONFIG_SND_DEBUG=y``. hwdep Lists the currently available hwdep devices in format of - <card>-<device>: <name> + ``<card>-<device>: <name>`` pcm Lists the currently available PCM devices in format of - <card>-<device>: <id>: <name> : <sub-streams> + ``<card>-<device>: <id>: <name> : <sub-streams>`` timer Lists the currently available timer devices @@ -54,23 +56,23 @@ oss/sndstat Card Specific Files -------------------- +=================== -The card-specific files are found in /proc/asound/card* directories. +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 +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 +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 +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#* +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), @@ -81,26 +83,27 @@ details. PCM Proc Files --------------- +============== -card*/pcm*/info +``card*/pcm*/info`` The general information of this PCM device: card #, device #, substreams, etc. -card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y and - CONFIG_PCM_XRUN_DEBUG=y. +``card*/pcm*/xrun_debug`` + This file appears when ``CONFIG_SND_DEBUG=y`` and + ``CONFIG_PCM_XRUN_DEBUG=y``. This shows the status of xrun (= buffer overrun/xrun) and invalid PCM position debug/check of ALSA PCM middle layer. It takes an integer value, can be changed by writing to this - file, such as + 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 + + * 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 @@ -117,72 +120,74 @@ card*/pcm*/xrun_debug buggy) hardware that doesn't give smooth pointer updates. This feature is enabled via the bit 2. -card*/pcm*/sub*/info +``card*/pcm*/sub*/info`` The general information of this PCM sub-stream. -card*/pcm*/sub*/status +``card*/pcm*/sub*/status`` The current status of this PCM sub-stream, elapsed time, H/W position, etc. -card*/pcm*/sub*/hw_params +``card*/pcm*/sub*/hw_params`` The hardware parameters set for this sub-stream. -card*/pcm*/sub*/sw_params +``card*/pcm*/sub*/sw_params`` The soft parameters set for this sub-stream. -card*/pcm*/sub*/prealloc +``card*/pcm*/sub*/prealloc`` The buffer pre-allocation information. -card*/pcm*/sub*/xrun_injection +``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#?-? +``card*/codec97#*/ac97#?-?`` Shows the general information of this AC97 codec chip, such as name, capabilities, set up. -card*/codec97#0/ac97#?-?+regs +``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* +``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#* +``card*/codec#*`` Shows the general codec information and the attribute of each widget node. -card*/eld#* +``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#*`. + 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. @@ -203,7 +208,7 @@ seq/oss 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 @@ -211,24 +216,23 @@ 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) +- 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 +- 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 +- AC97 proc files, ``codec97#*/*`` files for USB audio/midi, - - output of lsusb -v - - stream* files in card directory +- output of ``lsusb -v`` +- ``stream*`` files in card directory The ALSA bug-tracking system is found at: - - https://bugtrack.alsa-project.org/alsa-bug/ +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 000000000000..e82ffe0e7f43 --- /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/alsa/timestamping.txt b/Documentation/sound/designs/timestamping.rst index 9d579aefbffd..2b0fff503415 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/designs/timestamping.rst @@ -1,18 +1,22 @@ +===================== +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. + 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. + 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 @@ -22,15 +26,15 @@ 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. +- ``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 +``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 @@ -38,17 +42,18 @@ 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---->| ---------------------------------------------------------------> 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 @@ -113,11 +118,11 @@ 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 +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 +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. +``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 @@ -135,36 +140,42 @@ 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 +before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases this driver_timestamp will be identical to the regular system tstamp. Examples of typestamping with HDaudio: 1. DMA timestamp, no compensation for DMA+analog delay -$ ./audio_time -p --ts_type=1 -playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 -playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 -playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 -playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 -playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 -playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 +:: + + $ ./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 +:: + + $ ./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 +:: + + $ ./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 @@ -181,20 +192,24 @@ 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 +:: + + $ ./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 +:: + + $ ./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/alsa/HD-Audio-Controls.txt b/Documentation/sound/hd-audio/controls.rst index e9621e349e17..f2ebc4f79b44 100644 --- a/Documentation/sound/alsa/HD-Audio-Controls.txt +++ b/Documentation/sound/hd-audio/controls.rst @@ -1,16 +1,21 @@ +====================================== +HD-Audio Codec-Specific Mixer Controls +====================================== + + This file explains the codec-specific mixer controls. Realtek codecs -------------- -* Channel Mode +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 +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 @@ -30,24 +35,24 @@ Realtek codecs IDT/Sigmatel codecs ------------------- -* Analog Loopback +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 +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 +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 +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" @@ -57,19 +62,19 @@ IDT/Sigmatel codecs VIA codecs ---------- -* Smart 5.1 +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 +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 +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 @@ -78,7 +83,7 @@ VIA codecs headphones and speakers because there is only one DAC connected to a mixer widget. -* Dynamic Power-Control +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 @@ -86,7 +91,7 @@ VIA codecs doesn't provide a proper jack-detection, this won't work; in such a case, turn this control OFF. -* Jack Detect +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 @@ -96,21 +101,21 @@ VIA codecs Conexant codecs --------------- -* Auto-Mute Mode +Auto-Mute Mode See Reatek codecs. Analog codecs -------------- -* Channel Mode +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 +Independent HP When this enum control is enabled, the headphone output is routed from an individual stream (the third PCM such as hw:0,2) instead of the primary stream. diff --git a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt b/Documentation/sound/hd-audio/dp-mst.rst index 82744ac3513d..58b72437e6c3 100644 --- a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt +++ b/Documentation/sound/hd-audio/dp-mst.rst @@ -1,3 +1,7 @@ +======================= +HD-Audio DP-MST Support +======================= + To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin and dynamic pcm assignment. @@ -44,10 +48,12 @@ Build Jack ---------- - dyn_pcm_assign -Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly. + + 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. + + Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically. Unsolicited Event Enabling @@ -58,16 +64,20 @@ 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 + + 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. + + 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 000000000000..f8a72ffffe66 --- /dev/null +++ b/Documentation/sound/hd-audio/index.rst @@ -0,0 +1,10 @@ +HD-Audio +======== + +.. toctree:: + :maxdepth: 2 + + notes + models + controls + dp-mst diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst new file mode 100644 index 000000000000..5338673c88d9 --- /dev/null +++ b/Documentation/sound/hd-audio/models.rst @@ -0,0 +1,518 @@ +============================== +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 + +ALC260 +====== +gpio1 + Enable GPIO1 +coef + Enable EAPD via COEF table +fujitsu + Quirk for FSC S7020 +fujitsu-jwse + Quirk for FSC S7020 with jack modes and HP mic support + +ALC262 +====== +inv-dmic + Inverted internal mic workaround + +ALC267/268 +========== +inv-dmic + Inverted internal mic workaround +hp-eapd + Disable HP EAPD on NID 0x15 + +ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) +=================================================================== +laptop-amic + Laptops with analog-mic input +laptop-dmic + Laptops with digital-mic input +alc269-dmic + Enable ALC269(VA) digital mic workaround +alc271-dmic + Enable ALC271X digital mic workaround +inv-dmic + Inverted internal mic workaround +headset-mic + Indicates a combined headset (headphone+mic) jack +headset-mode + More comprehensive headset support for ALC269 & co +headset-mode-no-hp-mic + Headset mode support without headphone mic +lenovo-dock + Enables docking station I/O for some Lenovos +hp-gpio-led + GPIO LED support on HP laptops +dell-headset-multi + Headset jack, which can also be used as mic-in +dell-headset-dock + Headset jack (without mic-in), and also dock I/O +alc283-dac-wcaps + Fixups for Chromebook with ALC283 +alc283-sense-combo + Combo jack sensing on ALC283 +tpt440-dock + Pin configs for Lenovo Thinkpad Dock support + +ALC66x/67x/892 +============== +mario + Chromebook mario model fixup +asus-mode1 + ASUS +asus-mode2 + ASUS +asus-mode3 + ASUS +asus-mode4 + ASUS +asus-mode5 + ASUS +asus-mode6 + ASUS +asus-mode7 + ASUS +asus-mode8 + ASUS +inv-dmic + Inverted internal mic workaround +dell-headset-multi + Headset jack, which can also be used as mic-in + +ALC680 +====== +N/A + +ALC88x/898/1150 +====================== +acer-aspire-4930g + Acer Aspire 4930G/5930G/6530G/6930G/7730G +acer-aspire-8930g + Acer Aspire 8330G/6935G +acer-aspire + Acer Aspire others +inv-dmic + Inverted internal mic workaround +no-primary-hp + VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) + +ALC861/660 +========== +N/A + +ALC861VD/660VD +============== +N/A + +CMI9880 +======= +minimal + 3-jack in back +min_fp + 3-jack in back, 2-jack in front +full + 6-jack in back, 2-jack in front +full_dig + 6-jack in back, 2-jack in front, SPDIF I/O +allout + 5-jack in back, 2-jack in front, SPDIF out +auto + auto-config reading BIOS (default) + +AD1882 / AD1882A +================ +3stack + 3-stack mode +3stack-automute + 3-stack with automute front HP (default) +6stack + 6-stack mode + +AD1884A / AD1883 / AD1984A / AD1984B +==================================== +desktop 3-stack desktop (default) +laptop laptop with HP jack sensing +mobile mobile devices with HP jack sensing +thinkpad Lenovo Thinkpad X300 +touchsmart HP Touchsmart + +AD1884 +====== +N/A + +AD1981 +====== +basic 3-jack (default) +hp HP nx6320 +thinkpad Lenovo Thinkpad T60/X60/Z60 +toshiba Toshiba U205 + +AD1983 +====== +N/A + +AD1984 +====== +basic default configuration +thinkpad Lenovo Thinkpad T61/X61 +dell_desktop Dell T3400 + +AD1986A +======= +3stack + 3-stack, shared surrounds +laptop + 2-channel only (FSC V2060, Samsung M50) +laptop-imic + 2-channel with built-in mic +eapd + Turn on EAPD constantly + +AD1988/AD1988B/AD1989A/AD1989B +============================== +6stack + 6-jack +6stack-dig + ditto with SPDIF +3stack + 3-jack +3stack-dig + ditto with SPDIF +laptop + 3-jack with hp-jack automute +laptop-dig + ditto with SPDIF +auto + auto-config reading BIOS (default) + +Conexant 5045 +============= +laptop-hpsense + Laptop with HP sense (old model laptop) +laptop-micsense + Laptop with Mic sense (old model fujitsu) +laptop-hpmicsense + Laptop with HP and Mic senses +benq + Benq R55E +laptop-hp530 + HP 530 laptop +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5047 +============= +laptop + Basic Laptop config +laptop-hp + Laptop config for some HP models (subdevice 30A5) +laptop-eapd + Laptop config with EAPD support +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5051 +============= +laptop + Basic Laptop config (default) +hp + HP Spartan laptop +hp-dv6736 + HP dv6736 +hp-f700 + HP Compaq Presario F700 +ideapad + Lenovo IdeaPad laptop +toshiba + Toshiba Satellite M300 + +Conexant 5066 +============= +laptop + Basic Laptop config (default) +hp-laptop + HP laptops, e g G60 +asus + Asus K52JU, Lenovo G560 +dell-laptop + Dell laptops +dell-vostro + Dell Vostro +olpc-xo-1_5 + OLPC XO 1.5 +ideapad + Lenovo IdeaPad U150 +thinkpad + Lenovo Thinkpad + +STAC9200 +======== +ref + Reference board +oqo + OQO Model 2 +dell-d21 + Dell (unknown) +dell-d22 + Dell (unknown) +dell-d23 + Dell (unknown) +dell-m21 + Dell Inspiron 630m, Dell Inspiron 640m +dell-m22 + Dell Latitude D620, Dell Latitude D820 +dell-m23 + Dell XPS M1710, Dell Precision M90 +dell-m24 + Dell Latitude 120L +dell-m25 + Dell Inspiron E1505n +dell-m26 + Dell Inspiron 1501 +dell-m27 + Dell Inspiron E1705/9400 +gateway-m4 + Gateway laptops with EAPD control +gateway-m4-2 + Gateway laptops with EAPD control +panasonic + Panasonic CF-74 +auto + BIOS setup (default) + +STAC9205/9254 +============= +ref + Reference board +dell-m42 + Dell (unknown) +dell-m43 + Dell Precision +dell-m44 + Dell Inspiron +eapd + Keep EAPD on (e.g. Gateway T1616) +auto + BIOS setup (default) + +STAC9220/9221 +============= +ref + Reference board +3stack + D945 3stack +5stack + D945 5stack + SPDIF +intel-mac-v1 + Intel Mac Type 1 +intel-mac-v2 + Intel Mac Type 2 +intel-mac-v3 + Intel Mac Type 3 +intel-mac-v4 + Intel Mac Type 4 +intel-mac-v5 + Intel Mac Type 5 +intel-mac-auto + Intel Mac (detect type according to subsystem id) +macmini + Intel Mac Mini (equivalent with type 3) +macbook + Intel Mac Book (eq. type 5) +macbook-pro-v1 + Intel Mac Book Pro 1st generation (eq. type 3) +macbook-pro + Intel Mac Book Pro 2nd generation (eq. type 3) +imac-intel + Intel iMac (eq. type 2) +imac-intel-20 + Intel iMac (newer version) (eq. type 3) +ecs202 + ECS/PC chips +dell-d81 + Dell (unknown) +dell-d82 + Dell (unknown) +dell-m81 + Dell (unknown) +dell-m82 + Dell XPS M1210 +auto + BIOS setup (default) + +STAC9202/9250/9251 +================== +ref + Reference board, base config +m1 + Some Gateway MX series laptops (NX560XL) +m1-2 + Some Gateway MX series laptops (MX6453) +m2 + Some Gateway MX series laptops (M255) +m2-2 + Some Gateway MX series laptops +m3 + Some Gateway MX series laptops +m5 + Some Gateway MX series laptops (MP6954) +m6 + Some Gateway NX series laptops +auto + BIOS setup (default) + +STAC9227/9228/9229/927x +======================= +ref + Reference board +ref-no-jd + Reference board without HP/Mic jack detection +3stack + D965 3stack +5stack + D965 5stack + SPDIF +5stack-no-fp + D965 5stack without front panel +dell-3stack + Dell Dimension E520 +dell-bios + Fixes with Dell BIOS setup +dell-bios-amic + Fixes with Dell BIOS setup including analog mic +volknob + Fixes with volume-knob widget 0x24 +auto + BIOS setup (default) + +STAC92HD71B* +============ +ref + Reference board +dell-m4-1 + Dell desktops +dell-m4-2 + Dell desktops +dell-m4-3 + Dell desktops +hp-m4 + HP mini 1000 +hp-dv5 + HP dv series +hp-hdx + HP HDX series +hp-dv4-1222nr + HP dv4-1222nr (with LED support) +auto + BIOS setup (default) + +STAC92HD73* +=========== +ref + Reference board +no-jd + BIOS setup but without jack-detection +intel + Intel DG45* mobos +dell-m6-amic + Dell desktops/laptops with analog mics +dell-m6-dmic + Dell desktops/laptops with digital mics +dell-m6 + Dell desktops/laptops with both type of mics +dell-eq + Dell desktops/laptops +alienware + Alienware M17x +auto + BIOS setup (default) + +STAC92HD83* +=========== +ref + Reference board +mic-ref + Reference board with power management for ports +dell-s14 + Dell laptop +dell-vostro-3500 + Dell Vostro 3500 laptop +hp-dv7-4000 + HP dv-7 4000 +hp_cNB11_intquad + HP CNB models with 4 speakers +hp-zephyr + HP Zephyr +hp-led + HP with broken BIOS for mute LED +hp-inv-led + HP with broken BIOS for inverted mute LED +hp-mic-led + HP with mic-mute LED +headset-jack + Dell Latitude with a 4-pin headset jack +hp-envy-bass + Pin fixup for HP Envy bass speaker (NID 0x0f) +hp-envy-ts-bass + Pin fixup for HP Envy TS bass speaker (NID 0x10) +hp-bnb13-eq + Hardware equalizer setup for HP laptops +auto + BIOS setup (default) + +STAC92HD95 +========== +hp-led + LED support for HP laptops +hp-bass + Bass HPF setup for HP Spectre 13 + +STAC9872 +======== +vaio + VAIO laptop without SPDIF +auto + BIOS setup (default) + +Cirrus Logic CS4206/4207 +======================== +mbp55 + MacBook Pro 5,5 +imac27 + IMac 27 Inch +auto + BIOS setup (default) + +Cirrus Logic CS4208 +=================== +mba6 + MacBook Air 6,1 and 6,2 +gpio0 + Enable GPIO 0 amp +auto + BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== +auto + BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/hd-audio/notes.rst index d4510ebf2e8c..168d0cfab1ce 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/hd-audio/notes.rst @@ -1,10 +1,12 @@ -MORE NOTES ON HD-AUDIO DRIVER ============================= - Takashi Iwai <tiwai@suse.de> +More Notes on HD-Audio Driver +============================= +Takashi Iwai <tiwai@suse.de> -GENERAL -------- + +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 @@ -40,28 +42,28 @@ If you are interested in the deep debugging of HD-audio, read the HD-audio specification at first. The specification is found on Intel's web page, for example: -- http://www.intel.com/standards/hdaudio/ +* http://www.intel.com/standards/hdaudio/ -HD-AUDIO CONTROLLER -------------------- +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. +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 +``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, +``position_fix=4`` is another combination available for all controllers, and uses LPIB for the playback and the position-buffer for the capture streams. 0 is the default value for all other @@ -74,9 +76,9 @@ 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. +via ``bdl_pos_adj`` option. -When `bdl_pos_adj` is a negative value (as default), it's assigned to +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 @@ -84,19 +86,19 @@ 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 ------------------------------------------------------------------------- +:: + + 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 @@ -108,24 +110,24 @@ 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 +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. +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. +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 +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 @@ -134,17 +136,17 @@ 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 +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 --------------- +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 @@ -153,13 +155,15 @@ 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 @@ -182,32 +186,33 @@ 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 +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. +What ``model`` option values are available depends on the codec chip. Check your codec chip from the codec proc file (see "Codec Proc-File" section below). It will show the vendor/product name of your codec -chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file, +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 +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 +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 + +* 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 +* when ``generic`` is passed, the codec-specific parser is skipped and only the generic parser is used. Speaker and Headphone Output -~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +---------------------------- One of the most frequent (and obvious) bugs with HD-audio is the silent output from either or both of a built-in speaker and a headphone jack. In general, you should try a headphone output at @@ -236,23 +241,23 @@ 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 +* 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 +* 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 +* 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 +* 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 @@ -284,7 +289,7 @@ 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 @@ -293,45 +298,45 @@ below. You'd need to enable hwdep for using these tools. See "Kernel Configuration" section. -OTHER ISSUES ------------- +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. +``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 +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_*` +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. +``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 +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 +``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. +``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. @@ -351,161 +356,178 @@ 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. +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 ------------------------------------------------------------------------- +:: + + # 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 +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. +auto-switch. As a boolean value, either ``yes``, ``no``, ``true``, ``false``, +``1`` or ``0`` can be passed. The generic parser supports the following hints: -- jack_detect (bool): specify whether the jack detection is available - at all on this machine; default true -- inv_jack_detect (bool): indicates that the jack detection logic is - inverted -- trigger_sense (bool): indicates that the jack detection needs the - explicit call of AC_VERB_SET_PIN_SENSE verb -- inv_eapd (bool): indicates that the EAPD is implemented in the - inverted logic -- pcm_format_first (bool): sets the PCM format before the stream tag - and channel ID -- sticky_stream (bool): keep the PCM format, stream tag and ID as long - as possible; default true -- spdif_status_reset (bool): reset the SPDIF status bits at each time - the SPDIF stream is set up -- pin_amp_workaround (bool): the output pin may have multiple amp - values -- single_adc_amp (bool): ADCs can have only single input amps -- auto_mute (bool): enable/disable the headphone auto-mute feature; - default true -- auto_mic (bool): enable/disable the mic auto-switch feature; default - true -- line_in_auto_switch (bool): enable/disable the line-in auto-switch - feature; default false -- need_dac_fix (bool): limits the DACs depending on the channel count -- primary_hp (bool): probe headphone jacks as the primary outputs; - default true -- multi_io (bool): try probing multi-I/O config (e.g. shared - line-in/surround, mic/clfe jacks) -- multi_cap_vol (bool): provide multiple capture volumes -- inv_dmic_split (bool): provide split internal mic volume/switch for - phase-inverted digital mics -- indep_hp (bool): provide the independent headphone PCM stream and - the corresponding mixer control, if available -- add_stereo_mix_input (bool): add the stereo mix (analog-loopback - mix) to the input mux if available -- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each - I/O jack for allowing to change the headphone amp and mic bias VREF - capabilities -- power_save_node (bool): advanced power management for each widget, - controlling the power sate (D0/D3) of each widget node depending on - the actual pin and stream states -- power_down_unused (bool): power down the unused widgets, a subset of - power_save_node, and will be dropped in future -- add_hp_mic (bool): add the headphone to capture source if possible -- hp_mic_detect (bool): enable/disable the hp/mic shared input for a - single built-in mic case; default true -- mixer_nid (int): specifies the widget NID of the analog-loopback - mixer +jack_detect (bool) + specify whether the jack detection is available at all on this + machine; default true +inv_jack_detect (bool) + indicates that the jack detection logic is inverted +trigger_sense (bool) + indicates that the jack detection needs the explicit call of + AC_VERB_SET_PIN_SENSE verb +inv_eapd (bool) + indicates that the EAPD is implemented in the inverted logic +pcm_format_first (bool) + sets the PCM format before the stream tag and channel ID +sticky_stream (bool) + keep the PCM format, stream tag and ID as long as possible; + default true +spdif_status_reset (bool) + reset the SPDIF status bits at each time the SPDIF stream is set + up +pin_amp_workaround (bool) + the output pin may have multiple amp values +single_adc_amp (bool) + ADCs can have only single input amps +auto_mute (bool) + enable/disable the headphone auto-mute feature; default true +auto_mic (bool) + enable/disable the mic auto-switch feature; default true +line_in_auto_switch (bool) + enable/disable the line-in auto-switch feature; default false +need_dac_fix (bool) + limits the DACs depending on the channel count +primary_hp (bool) + probe headphone jacks as the primary outputs; default true +multi_io (bool) + try probing multi-I/O config (e.g. shared line-in/surround, + mic/clfe jacks) +multi_cap_vol (bool) + provide multiple capture volumes +inv_dmic_split (bool) + provide split internal mic volume/switch for phase-inverted + digital mics +indep_hp (bool) + provide the independent headphone PCM stream and the corresponding + mixer control, if available +add_stereo_mix_input (bool) + add the stereo mix (analog-loopback mix) to the input mux if + available +add_jack_modes (bool) + add "xxx Jack Mode" enum controls to each I/O jack for allowing to + change the headphone amp and mic bias VREF capabilities +power_save_node (bool) + advanced power management for each widget, controlling the power + sate (D0/D3) of each widget node depending on the actual pin and + stream states +power_down_unused (bool) + power down the unused widgets, a subset of power_save_node, and + will be dropped in future +add_hp_mic (bool) + add the headphone to capture source if possible +hp_mic_detect (bool) + enable/disable the hp/mic shared input for a single built-in mic + case; default true +mixer_nid (int) + specifies the widget NID of the analog-loopback mixer Early Patching -~~~~~~~~~~~~~~ -When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a -firmware file for modifying the HD-audio setup before initializing the -codec. This can work basically like the reconfiguration via sysfs in -the above, but it does it before the first codec configuration. +-------------- +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 +:: + + [codec] + 0x12345678 0xabcd1234 2 - [model] - auto + [model] + auto - [pincfg] - 0x12 0x411111f0 + [pincfg] + 0x12 0x411111f0 - [verb] - 0x20 0x500 0x03 - 0x20 0x400 0xff + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff - [hint] - jack_detect = no ------------------------------------------------------------------------- + [hint] + jack_detect = no -The file needs to have a line `[codec]`. The next line should contain + +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 @@ -514,66 +536,68 @@ 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. +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. +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` +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 +:: + + [codec] + 0x12345678 0xabcd1234 2 + + [vendor_id] + 0xdeadbeef - [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]``, 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 - [subsystem_id] - 0xffff1111 + [revision_id] + 0x10 - [revision_id] - 0x10 + [chip_name] + My-own NEWS-0002 - [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 +``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 +``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 @@ -592,63 +616,65 @@ 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 +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` +``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 +``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 ------------------------------------------------------------------------- +:: + + # 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 +:: + + # 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 ------------------------------------------------------------------------- +:: + + % 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 +* 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 @@ -657,14 +683,14 @@ 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 +* 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 @@ -673,60 +699,60 @@ compare the codec registers directly. Send a bug report either the followings: -kernel-bugzilla:: - https://bugzilla.kernel.org/ -alsa-devel ML:: - alsa-devel@alsa-project.org +kernel-bugzilla + https://bugzilla.kernel.org/ +alsa-devel ML + alsa-devel@alsa-project.org -DEBUG TOOLS ------------ +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 +--------- +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 +* git://git.alsa-project.org/alsa-utils.git The script can be fetched directly from the following URL, too: -- http://www.alsa-project.org/alsa-info.sh +* http://www.alsa-project.org/alsa-info.sh Run this script as root, and it will gather the important information such as the module lists, module parameters, proc file contents including the codec proc files, mixer outputs and the control elements. As default, it will store the information onto a web server on alsa-project.org. But, if you send a bug report, it'd be better to -run with `--no-upload` option, and attach the generated file. +run with ``--no-upload`` option, and attach the generated file. -There are some other useful options. See `--help` option output for +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 +``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 +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 +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. +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 @@ -739,19 +765,20 @@ 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 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 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 - % 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 @@ -760,22 +787,22 @@ 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 +* 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/ +* 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 +* 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 @@ -784,14 +811,14 @@ proc-compatible output. The hda-analyzer: -- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer +* http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer is a part of alsa.git repository in alsa-project.org: -- git://git.alsa-project.org/alsa.git +* 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 @@ -800,11 +827,11 @@ program. The tarball and GIT trees are found in the web page at: -- http://helllabs.org/codecgraph/ +* 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 @@ -817,13 +844,14 @@ 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 - .... ------------------------------------------------------------------------- +:: + + % 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 @@ -832,14 +860,14 @@ 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 +* 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 @@ -849,5 +877,4 @@ firmware patch file (see "Early Patching" section). The program is included in alsa-tools now: -- git://git.alsa-project.org/alsa-tools.git - +* git://git.alsa-project.org/alsa-tools.git diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst new file mode 100644 index 000000000000..47b89f014e69 --- /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 + + 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 000000000000..14cd138989e3 --- /dev/null +++ b/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -0,0 +1,134 @@ +=================== +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-io.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 diff --git a/Documentation/sound/kernel-api/index.rst b/Documentation/sound/kernel-api/index.rst new file mode 100644 index 000000000000..d0e6df35b4b4 --- /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 000000000000..95c5443eff38 --- /dev/null +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -0,0 +1,4219 @@ +====================== +Writing an ALSA Driver +====================== + +:Author: Takashi Iwai <tiwai@suse.de> +:Date: Oct 15, 2007 +:Edition: 0.3.7 + +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. + +If you are already familiar with the older ALSA ver.0.5.x API, you can +check the drivers such as ``sound/pci/es1938.c`` or +``sound/pci/maestro3.c`` which have also almost the same code-base in +the ALSA 0.5.x tree, so you can compare the differences. + +This document is still a draft version. Any feedback and corrections, +please!! + +File Tree Structure +=================== + +General +------- + +The ALSA drivers are provided in two ways. + +One is the trees provided as a tarball or via cvs from the ALSA's ftp +site, and another is the 2.6 (or later) Linux kernel tree. To +synchronize both, the ALSA driver tree is split into two different +trees: alsa-kernel and alsa-driver. The former contains purely the +source code for the Linux 2.6 (or later) tree. This tree is designed +only for compilation on 2.6 or later environment. The latter, +alsa-driver, contains many subtle files for compiling ALSA drivers +outside of the Linux kernel tree, wrapper functions for older 2.2 and +2.4 kernels, to adapt the latest kernel API, and additional drivers +which are still in development or in tests. The drivers in alsa-driver +tree will be moved to alsa-kernel (and eventually to the 2.6 kernel +tree) when they are finished and confirmed to work fine. + +The file tree structure of ALSA driver is depicted below. Both +alsa-kernel and alsa-driver have almost the same file structure, except +for “core” directory. It's named as “acore” in alsa-driver tree. + +:: + + sound + /core + /oss + /seq + /oss + /instr + /ioctl32 + /include + /drivers + /mpu401 + /opl3 + /i2c + /l3 + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /oss + + +core directory +-------------- + +This directory contains the middle layer 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/ioctl32 +~~~~~~~~~~~~ + +This directory contains the 32bit-ioctl wrappers for 64bit architectures +such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, +these are not compiled. + +core/seq +~~~~~~~~ + +This 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. + +core/seq/instr +~~~~~~~~~~~~~~ + +This directory contains the modules for the sequencer instrument layer. + +include directory +----------------- + +This is the place for the public header files of ALSA drivers, which are +to be exported to 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. + +i2c/l3 +~~~~~~ + +This is a sub-directory for ARM L3 i2c. + +synth directory +--------------- + +This contains the synth middle-level modules. + +So far, there is only Emu8000/Emu10k1 synth driver under 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. + +oss directory +------------- + +The OSS/Lite source files are stored here in Linux 2.6 (or later) tree. +In the ALSA driver tarball, this directory is empty, of course :) + +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 :c:type:`struct pci_driver <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 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) { + snd_card_free(card); + return err; + } + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see the "Destructor" sub-section */ + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + + +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 +<#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) { + snd_card_free(card); + return err; + } + +The details will be explained in the section `PCI Resource +Management`_. + +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->ioport, 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) { + snd_card_free(card); + return err; + } + +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 like the following: + +:: + + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + +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 ``<asm/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 :c:type:`struct device +<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 +:c:type:`struct snd_device <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); + +:c:type:`struct mychip <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 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. + +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 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; + + /* (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. Unlike ALSA ver.0.5.x., there are no helpers for that. And +these resources must be released in the destructor function (see below). +Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI +like in ALSA 0.5.x. + +Now assume that the PCI device has an I/O port with 8 bytes and an +interrupt. Then :c:type:`struct mychip <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-interface-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; + } + + +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: + +:: + + if ((err = pci_request_regions(pci, "My Chip")) < 0) { + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); + +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); + .... + } + +PCI Entries +----------- + +So far, so good. Let's finish the missing PCI stuff. At first, we need a +:c:type:`struct pci_device_id <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 :c:type:`struct pci_device_id +<pci_device_id>` structure are the vendor and device IDs. If you +have no reason to filter the matching devices, you can leave the +remaining fields as above. The last field of the :c:type:`struct +pci_device_id <pci_device_id>` struct contains private data +for this entry. You can specify any value here, for example, to define +specific operations for supported device IDs. Such an example is found +in the intel8x0 driver. + +The last entry of this list is the terminator. You must specify this +all-zero entry. + +Then, prepare the :c:type:`struct pci_driver <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. + +Oh, one thing was forgotten. If you have no exported symbols, you need +to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). + +:: + + EXPORT_NO_SYMBOLS; + +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) + { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + return 0; + } + + +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 :c:type:`struct snd_pcm_substream +<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, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + +All the callbacks are described in the Operators_ subsection. + +After setting the operators, you probably will want to pre-allocate the +buffer. For the pre-allocation, simply call the following: + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 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 buffer allocation method via +:c:func:`snd_pcm_lib_malloc_pages()`, you don't need to set the +DMA buffer information by yourself. + +In the sections below, important records are explained. + +Hardware Description +~~~~~~~~~~~~~~~~~~~~ + +The hardware descriptor (:c:type:`struct snd_pcm_hardware +<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 a standard ALSA 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 :c:type:`struct snd_pcm_mmap_status +<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 :c:type:`struct snd_pcm_mmap_control +<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 :c:type:`struct +snd_pcm_substream <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 +pass a generic ioctl callback, :c:func:`snd_pcm_lib_ioctl()`. + +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. To allocate buffer, you can call a +helper function, + +:: + + 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``. For example, releasing the buffer via +:c:func:`snd_pcm_lib_malloc_pages()` is done by calling the +following: + +:: + + snd_pcm_lib_free_pages(substream); + +This function is always called before the close callback is called. +Also, the callback may be called multiple times, too. Keep track +whether the resource was already released. + +prepare callback +~~~~~~~~~~~~~~~~ + +:: + + 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. + +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 and silence callbacks +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +These callbacks are not mandatory, and can be omitted in most cases. +These callbacks are used when the hardware buffer cannot be 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. This callback is used mainly for +non-contiguous buffers. The mmap calls this callback to get the page +address. Some examples will be explained in the later section `Buffer +and Memory Management`_, too. + +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 :c:type:`struct snd_pcm +<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 the :c:type:`struct snd_pcm_hardware +<snd_pcm_hardware>` structure (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); + +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 +:c:type:`struct snd_kcontrol_new <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/alsa/ControlNames.txt``. + +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 :c:type:`struct +snd_ctl_elem_info <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 :c:type:`struct snd_kcontrol_new +<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 :c:type:`struct +snd_kcontrol <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 :c:type:`struct +snd_ctl_elem_id <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 :c:type:`struct +snd_ac97_template <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 :c:type:`struct snd_mpu401 <snd_mpu401>` manually afterward. +However, :c:type:`struct snd_mpu401 <snd_mpu401>` pointer is +not returned explicitly by :c:func:`snd_mpu401_uart_new()`. You +need to cast ``rmidi->private_data`` to :c:type:`struct snd_mpu401 +<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 +:c:func:`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, +and are defined as MIXER controls (iface = +``SNDRV_CTL_ELEM_IFACE_MIXER``). + +Meanwhile, “IEC958 Playback Default” control is defined for getting and +setting the current default IEC958 bits. Note that this one is usually +defined as a PCM control (iface = ``SNDRV_CTL_ELEM_IFACE_PCM``), +although in some places it's defined as a MIXER control. + +In addition, you can define the control switches to enable/disable or to +set the raw bit mode. The implementation will depend on the chip, but +the control should be named as “IEC958 xxx”, preferably using 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, + snd_dma_pci_data(pci), 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. In the case of the ISA bus, pass +:c:func:`snd_dma_isa_data()` as the third argument with +``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the +bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type and the +``snd_dma_continuous_data(GFP_KERNEL)`` device pointer, where +``GFP_KERNEL`` is the kernel allocation flag to use. For the PCI +scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with +``snd_dma_pci_data(pci)`` (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. + +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`` and ``silence`` callbacks +for the data transfer. However, there is a drawback: it cannot be +mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. + +The second case allows 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`` and ``silence`` callbacks as well, as in the cases above. The +examples are found in ``rme32.c`` and ``rme96.c``. + +The implementation of the ``copy`` and ``silence`` callbacks depends +upon whether the hardware supports interleaved or non-interleaved +samples. The ``copy`` callback is defined like below, a bit +differently depending whether the direction is playback or capture: + +:: + + static int playback_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); + static int capture_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); + +In the case of interleaved samples, the second argument (``channel``) is +not used. The third argument (``pos``) points the current position +offset in frames. + +The meaning of the fourth argument is different between playback and +capture. For playback, it holds the source data pointer, and for +capture, it's the destination data pointer. + +The last argument is the number of frames to be copied. + +What you have to do in this callback is again different between playback +and capture directions. In the 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(my_buffer + frames_to_bytes(runtime, pos), src, + frames_to_bytes(runtime, 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(dst, my_buffer + frames_to_bytes(runtime, pos), + frames_to_bytes(runtime, count)); + +Note that both the position and the amount of data are given in frames. + +In the case of non-interleaved samples, the implementation will be a bit +more complicated. + +You need to check the channel argument, and if it's -1, copy the whole +channels. Otherwise, you have to copy only the specified channel. Please +check ``isa/gus/gus_pcm.c`` as an example. + +The ``silence`` callback is also implemented in a similar way + +:: + + static int silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count); + +The meanings of arguments are the same as in the ``copy`` callback, +although there is no ``src/dst`` argument. In the case of interleaved +samples, the channel argument has no meaning, as well as on ``copy`` +callback. + +The role of ``silence`` callback is to set the given amount +(``count``) of silence data at the specified offset (``pos``) on the +hardware buffer. Suppose that the data format is signed (that is, the +silent-data is 0), and the implementation using a memset-like function +would be like: + +:: + + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, + frames_to_bytes(runtime, count)); + +In the case of non-interleaved samples, again, the implementation +becomes a bit more complicated. See, for example, ``isa/gus/gus_pcm.c``. + +Non-Contiguous Buffers +---------------------- + +If your hardware supports the page table 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_lib_preallocate_pages()` or +:c:func:`snd_pcm_lib_preallocate_pages_for_all()` with +``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI +pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is +the :c:type:`struct pci_dev <pci_dev>` pointer of the chip as +well. The ``struct snd_sg_buf`` instance is created as +``substream->dma_private``. You can cast the pointer like: + +:: + + struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; + +Then call :c:func:`snd_pcm_lib_malloc_pages()` in the ``hw_params`` +callback as well as in the case of normal PCI buffer. The SG-buffer +handler will allocate the non-contiguous kernel pages of the given size +and map them onto the virtually contiguous memory. The virtual pointer +is addressed in runtime->dma_area. The physical address +(``runtime->dma_addr``) is set to zero, because the buffer is +physically non-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()`. + +When a SG-handler is used, you need to set +:c:func:`snd_pcm_sgbuf_ops_page()` as the ``page`` callback. (See +`page callback`_ section.) + +To release the data, call :c:func:`snd_pcm_lib_free_pages()` in +the ``hw_free`` callback as usual. + +Vmalloc'ed Buffers +------------------ + +It's possible to use a buffer allocated via :c:func:`vmalloc()`, for +example, for an intermediate buffer. Since the allocated pages are not +contiguous, you need to set the ``page`` callback to obtain the physical +address at every offset. + +The implementation of ``page`` callback would be like this: + +:: + + #include <linux/vmalloc.h> + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(struct snd_pcm_substream *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } + +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 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/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``. + +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: + +:: + + #ifdef CONFIG_PM + static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + { + .... /* do things for suspend */ + return 0; + } + static int snd_my_resume(struct pci_dev *pci) + { + .... /* do things for suspend */ + return 0; + } + #endif + +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. Call :c:func:`snd_pcm_suspend_all()` to suspend the running + PCM streams. + +4. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for + each codec. + +5. Save the register values if necessary. + +6. Stop the hardware if necessary. + +7. Disable the PCI device by calling + :c:func:`pci_disable_device()`. Then, call + :c:func:`pci_save_state()` at last. + +A typical code would be like: + +:: + + static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_pcm_suspend_all(chip->pcm); + /* (4) */ + snd_ac97_suspend(chip->ac97); + /* (5) */ + snd_mychip_save_registers(chip); + /* (6) */ + snd_mychip_stop_hardware(chip); + /* (7) */ + pci_disable_device(pci); + pci_save_state(pci); + return 0; + } + + +The scheme of the real resume job is as follows. + +1. Retrieve the card and the chip data. + +2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then + enable the pci device again by calling + :c:func:`pci_enable_device()`. Call + :c:func:`pci_set_master()` if necessary, too. + +3. Re-initialize the chip. + +4. Restore the saved registers if necessary. + +5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. + +6. Restart the hardware (if any). + +7. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D0`` to notify the processes. + +A typical code would be like: + +:: + + static int mychip_resume(struct pci_dev *pci) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + /* (3) */ + snd_mychip_reinit_chip(chip); + /* (4) */ + snd_mychip_restore_registers(chip); + /* (5) */ + snd_ac97_resume(chip->ac97); + /* (6) */ + snd_mychip_restart_chip(chip); + /* (7) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } + +As shown in the above, it's better to save registers after suspending +the PCM operations via :c:func:`snd_pcm_suspend_all()` or +:c:func:`snd_pcm_suspend()`. It means that the PCM streams are +already stopped when the register snapshot is taken. But, remember that +you don't have to restart the PCM stream in the resume callback. It'll +be restarted via trigger call with ``SNDRV_PCM_TRIGGER_RESUME`` when +necessary. + +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 struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_my_ids, + .probe = snd_my_probe, + .remove = snd_my_remove, + #ifdef CONFIG_PM + .suspend = snd_my_suspend, + .resume = snd_my_resume, + #endif + }; + +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, classes, license +and devices. Especially, the recent modprobe requires to define the +module license as GPL, etc., otherwise the system is shown as “tainted”. + +:: + + MODULE_DESCRIPTION("My Chip"); + MODULE_LICENSE("GPL"); + MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); + + +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, ``alsa-driver/pci`` directory in the case of PCI +cards. Then the driver is evaluated, audited and tested by developers +and users. After a certain time, the driver will go to the alsa-kernel +tree (to the corresponding directory, such as ``alsa-kernel/pci``) and +eventually will be integrated into the Linux 2.6 tree (the directory +would be ``linux/sound/pci``). + +In the following sections, the driver code is supposed to be put into +alsa-driver tree. The two cases are covered: a driver consisting of a +single source file and one consisting of several source files. + +Driver with A Single Source File +-------------------------------- + +1. Modify alsa-driver/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. + +3. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Drivers with Several Source Files +--------------------------------- + +Suppose that the driver snd-xyz have several source files. They are +located in the new subdirectory, pci/xyz. + +1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as + below + +:: + + obj-$(CONFIG_SND) += xyz/ + + +2. Under the directory ``xyz``, create a Makefile + +:: + + ifndef SND_TOPDIR + SND_TOPDIR=../.. + endif + + include $(SND_TOPDIR)/toplevel.config + include $(SND_TOPDIR)/Makefile.conf + + snd-xyz-objs := xyz.o abc.o def.o + + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + + include $(SND_TOPDIR)/Rules.make + +3. Create the Kconfig entry + + This procedure is as same as in the last section. + +4. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Useful Functions +================ + +:c:func:`snd_printk()` and friends +--------------------------------------- + +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. Please note that +``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure +the alsa-driver with ``--with-debug=full`` option. You need to give +explicitly ``--with-debug=detect`` option instead. + +: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/alsa/soc/clocking.txt b/Documentation/sound/soc/clocking.rst index b1300162e01c..32122d6877a3 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/soc/clocking.rst @@ -1,3 +1,4 @@ +============== Audio Clocking ============== @@ -30,15 +31,9 @@ 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 +- 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 diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst new file mode 100644 index 000000000000..810109d7500d --- /dev/null +++ b/Documentation/sound/soc/codec-to-codec.rst @@ -0,0 +1,108 @@ +============================================== +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. + +Note that in current device tree there is no way to mark a dai_link +as codec to codec. However, it may change in future. diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/soc/codec.rst index db5f9c9ae149..f87612b94812 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/soc/codec.rst @@ -1,3 +1,4 @@ +======================= ASoC Codec Class Driver ======================= @@ -9,16 +10,16 @@ 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. +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. +7. DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec driver code in sound/soc/codecs/ @@ -26,24 +27,25 @@ driver code in sound/soc/codecs/ ASoC Codec driver breakdown =========================== -1 - Codec DAI and PCM configuration ------------------------------------ +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 = { + static struct snd_soc_dai_ops wm8731_dai_ops = { .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -}; - -struct snd_soc_dai_driver wm8731_dai = { + }; + + struct snd_soc_dai_driver wm8731_dai = { .name = "wm8731-hifi", .playback = { .stream_name = "Playback", @@ -59,25 +61,27 @@ struct snd_soc_dai_driver wm8731_dai = { .formats = WM8731_FORMATS,}, .ops = &wm8731_dai_ops, .symmetric_rates = 1, -}; + }; -2 - Codec control IO --------------------- +Codec control IO +---------------- The codec can usually be controlled via an I2C or SPI style interface (AC97 combines control with data in the DAI). The codec driver should use the Regmap API for all codec IO. Please see include/linux/regmap.h and existing codec drivers for example regmap usage. -3 - Mixers and audio controls ------------------------------ +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 @@ -86,18 +90,22 @@ Defines a single control as follows:- 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 @@ -109,25 +117,26 @@ Defines an single enumerated control as follows:- Defines a stereo enumerated control -4 - Codec Audio Operations --------------------------- +Codec Audio Operations +---------------------- The codec driver also supports the following ALSA PCM operations:- +:: -/* SoC audio ops */ -struct snd_soc_ops { + /* SoC audio ops */ + struct snd_soc_ops { int (*startup)(struct snd_pcm_substream *); void (*shutdown)(struct snd_pcm_substream *); int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); -}; + }; Please refer to the ALSA driver PCM documentation for details. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ -5 - DAPM description. ---------------------- +DAPM description +---------------- The Dynamic Audio Power Management description describes the codec power components and their relationships and registers to the ASoC core. Please read dapm.txt for details of building the description. @@ -135,13 +144,14 @@ Please read dapm.txt for details of building the description. Please also see the examples in other codec drivers. -6 - DAPM event handler ----------------------- +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 */ @@ -155,8 +165,8 @@ Power states:- SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ -7 - Codec DAC digital mute control ----------------------------------- +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. @@ -165,9 +175,10 @@ 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) -{ + static int wm8974_mute(struct snd_soc_dai *dai, int mute) + { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; @@ -176,4 +187,4 @@ static int wm8974_mute(struct snd_soc_dai *dai, int mute) else snd_soc_write(codec, WM8974_DAC, mute_reg); return 0; -} + } diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/soc/dai.rst index c9679264c559..55820e51708f 100644 --- a/Documentation/sound/alsa/soc/DAI.txt +++ b/Documentation/sound/soc/dai.rst @@ -1,3 +1,7 @@ +================================== +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. @@ -5,21 +9,21 @@ 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 +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 :- +The AC97 specification can be found at : http://www.intel.com/p/en_US/business/design I2S === - I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and +I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and Rx lines are used for audio transmission, whilst the bit clock (BCLK) and left/right clock (LRC) synchronise the link. I2S is flexible in that either the controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock @@ -30,13 +34,15 @@ different sample rates. I2S has several different operating modes:- - o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC - transition. +I2S + MSB is transmitted on the falling edge of the first BCLK after LRC + transition. - o Left Justified - MSB is transmitted on transition of LRC. +Left Justified + MSB is transmitted on transition of LRC. - o Right Justified - MSB is transmitted sample size BCLKs before LRC - transition. +Right Justified + MSB is transmitted sample size BCLKs before LRC transition. PCM === @@ -51,6 +57,8 @@ is sometimes referred to as network mode). Common PCM operating modes:- - o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. +Mode A + MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. - o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. +Mode B + MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/soc/dapm.rst index c45bd79f291e..a27f42befa4d 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/soc/dapm.rst @@ -1,8 +1,9 @@ +=================================================== Dynamic Audio Power Management for Portable Devices =================================================== -1. Description -============== +Description +=========== Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices to use the minimum amount of power within the audio @@ -21,20 +22,28 @@ level power systems. There are 4 power domains within DAPM - 1. Codec bias domain - VREF, VMID (core codec and audio power) +Codec bias domain + VREF, VMID (core codec and audio power) + Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. - 2. Platform/Machine domain - physically connected inputs and outputs +Platform/Machine domain + physically connected inputs and outputs + Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted - 3. Path domain - audio subsystem signal paths +Path domain + audio subsystem signal paths + Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. - 4. Stream domain - DACs and ADCs. +Stream domain + DACs and ADCs. + Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. @@ -45,34 +54,57 @@ internal codec components). All audio components that effect power are called widgets hereafter. -2. DAPM Widgets -=============== +DAPM Widgets +============ Audio DAPM widgets fall into a number of types:- - o Mixer - Mixes several analog signals into a single analog signal. - o Mux - An analog switch that outputs only one of many inputs. - o PGA - A programmable gain amplifier or attenuation widget. - o ADC - Analog to Digital Converter - o DAC - Digital to Analog Converter - o Switch - An analog switch - o Input - A codec input pin - o Output - A codec output pin - o Headphone - Headphone (and optional Jack) - o Mic - Mic (and optional Jack) - o Line - Line Input/Output (and optional Jack) - o Speaker - Speaker - o Supply - Power or clock supply widget used by other widgets. - o Regulator - External regulator that supplies power to audio components. - o Clock - External clock that supplies clock to audio components. - o AIF IN - Audio Interface Input (with TDM slot mask). - o AIF OUT - Audio Interface Output (with TDM slot mask). - o Siggen - Signal Generator. - o DAI IN - Digital Audio Interface Input. - o DAI OUT - Digital Audio Interface Output. - o DAI Link - DAI Link between two DAI structures */ - o Pre - Special PRE widget (exec before all others) - o Post - Special POST widget (exec after all others) +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) (Widgets are defined in include/sound/soc-dapm.h) @@ -84,52 +116,57 @@ Most widgets have a name, register, shift and invert. Some widgets have extra parameters for stream name and kcontrols. -2.1 Stream Domain Widgets -------------------------- +Stream 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) + 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), + 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), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), -2.2 Path Domain Widgets ------------------------ +Path Domain Widgets +------------------- 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) + 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), -}; + /* 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, + 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, @@ -137,48 +174,49 @@ you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. -2.3 Machine domain Widgets --------------------------- +Machine domain Widgets +---------------------- Machine widgets are different from codec widgets in that they don't have a codec register bit associated with them. A machine widget is assigned to each machine audio component (non codec or DSP) that can be independently powered. e.g. - o Speaker Amp - o Microphone Bias - o Jack connectors +* 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:- +when the Mic is inserted:-:: -static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) -{ + 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), + SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), -2.4 Codec (BIAS) Domain ------------------------ +Codec (BIAS) Domain +------------------- The codec bias power domain has no widgets and is handled by the codecs DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. -2.5 Virtual Widgets -------------------- +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), + 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. @@ -186,8 +224,8 @@ After all the widgets have been defined, they can then be added to the DAPM subsystem individually with a call to snd_soc_dapm_new_control(). -3. Codec/DSP Widget Interconnections -==================================== +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 @@ -201,13 +239,14 @@ 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 +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"}, @@ -216,22 +255,17 @@ connect the destination widget (wrt audio signal) with its source widgets. 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". +* 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); + 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 @@ -239,12 +273,13 @@ scan the codec and machine so that the internal DAPM state matches the physical state of the machine. -3.1 Machine Widget Interconnections ------------------------------------ +Machine Widget Interconnections +------------------------------- 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"}, @@ -254,52 +289,54 @@ This allows the DAPM to power on and off pins that are connected (and in use) and pins that are NC respectively. -4 Endpoint Widgets -=================== +Endpoint Widgets +================ An endpoint is a start or end point (widget) of an audio signal within the machine and includes the codec. e.g. - o Headphone Jack - o Internal Speaker - o Internal Mic - o Mic Jack - o Codec Pins +* 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. -5 DAPM Widget Events -==================== +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) -{ + /* 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 = + /* corgi machine dapm widgets */ + static const struct snd_soc_dapm_widget wm8731_dapm_widgets = SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); Please see soc-dapm.h for all other widgets that support events. -5.1 Event types ---------------- +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 */ +:: + + /* dapm event types */ + #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ + #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ + #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ + #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ + #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ + #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/soc/dpcm.rst index 0110180b7ac6..395e5a516282 100644 --- a/Documentation/sound/alsa/soc/DPCM.txt +++ b/Documentation/sound/soc/dpcm.rst @@ -1,8 +1,9 @@ +=========== Dynamic PCM =========== -1. Description -============== +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 @@ -23,22 +24,23 @@ 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 - ************* +:: + + | 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 @@ -55,50 +57,52 @@ 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 - ************* +:: + + ************* + 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 - ************* +:: + + ************* + 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. +1. Machine driver receives Jack removal event. - 2) Machine driver OR audio HAL disables the Headset path. +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. +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. +4. Machine driver or audio HAL enables the speaker path. - 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and - trigger(start) for DAI1 Speakers since the path is enabled. +5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() and + trigger(start) for DAI1 Speakers since the path is enabled. In this example, the machine driver or userspace audio HAL can alter the routing and then DPCM will take care of managing the DAI PCM operations to either bring @@ -112,36 +116,38 @@ 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. +1. Define the FE and BE DAI links. - 2) Define any FE/BE PCM operations. +2. Define any FE/BE PCM operations. - 3) Define widget graph connections. +3. Define widget graph connections. -1 FE and BE DAI links ---------------------- +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 - ************* + | 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[] = { + static struct snd_soc_dai_link machine_dais[] = { { .name = "PCM0 System", .stream_name = "System Playback", @@ -154,11 +160,11 @@ static struct snd_soc_dai_link machine_dais[] = { .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" +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 @@ -168,8 +174,9 @@ 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[] = { + static struct snd_soc_dai_link machine_dais[] = { .....< FE DAI links here > { .name = "Codec Headset", @@ -186,29 +193,30 @@ static struct snd_soc_dai_link machine_dais[] = { .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 ``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 - ************* +:: + + ************* + PCM0 <------------> * * <----DAI0-----> Codec Headset + * * + PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * + PCM2 <------------> * * <====DAI2=====> MODEM + * * + PCM3 <------------> * * <====DAI3=====> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are still in operation. @@ -220,10 +228,10 @@ Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the DSP firmware. -2 FE/BE PCM operations ----------------------- +FE/BE PCM operations +-------------------- -The BE above also exports some PCM operations and a "fixup" callback. The fixup +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. @@ -231,10 +239,11 @@ e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo fo 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, + 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, @@ -249,21 +258,22 @@ static int dai0_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_FIRST_MASK], SNDRV_PCM_FORMAT_S16_LE); return 0; -} + } The other PCM operation are the same as for regular DAI links. Use as necessary. -3 Widget graph connections --------------------------- +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"}, + /* 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 @@ -273,24 +283,25 @@ 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. +1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver. - 2) DAPM graph showing DSP audio routing from FE DAIs to BEs. +2. DAPM graph showing DSP audio routing from FE DAIs to BEs. - 3) DAPM widgets from DSP graph. +3. DAPM widgets from DSP graph. - 4) Mixers for gains, routing, etc. +4. Mixers for gains, routing, etc. - 5) DMA configuration. +5. DMA configuration. - 6) BE AIF widgets. +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), + 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). @@ -301,33 +312,33 @@ 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 - ************* +:: + + ************* + 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 + 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 + 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 @@ -339,16 +350,17 @@ 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 = { + 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[] = { + static struct snd_soc_dai_link dais[] = { < ... more DAI links above ... > { .name = "MODEM", diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst new file mode 100644 index 000000000000..e57df2dab2fd --- /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/alsa/soc/jack.txt b/Documentation/sound/soc/jack.rst index fcf82a417293..644b99ecba35 100644 --- a/Documentation/sound/alsa/soc/jack.txt +++ b/Documentation/sound/soc/jack.rst @@ -1,3 +1,4 @@ +=================== ASoC jack detection =================== diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/soc/machine.rst index 6bf2d2063b52..515c9444deaf 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/soc/machine.rst @@ -1,3 +1,4 @@ +=================== ASoC Machine Driver =================== @@ -9,9 +10,10 @@ 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 { + /* SoC machine */ + struct snd_soc_card { char *name; ... @@ -33,7 +35,7 @@ struct snd_soc_card { int num_links; ... -}; + }; probe()/remove() ---------------- @@ -55,9 +57,10 @@ 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 = { + /* 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", @@ -66,16 +69,17 @@ static struct snd_soc_dai_link corgi_dai = { .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 = { + /* corgi audio machine driver */ + static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", .dai_link = &corgi_dai, .num_links = 1, -}; + }; Machine Power Map diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/soc/overview.rst index f3f28b7ae242..dc8370bbfff6 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/soc/overview.rst @@ -1,5 +1,6 @@ -ALSA SoC Layer -============== +======================= +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. @@ -66,30 +67,3 @@ multiple re-usable component drivers :- describes and binds the other component drivers together to form an ALSA "sound card device". It handles any machine specific controls and machine level audio events (e.g. turning on an amp at start of playback). - - -Documentation -============= - -The documentation is spilt into the following sections:- - -overview.txt: This file. - -codec.txt: Codec driver internals. - -DAI.txt: Description of Digital Audio Interface standards and how to configure -a DAI within your codec and CPU DAI drivers. - -dapm.txt: Dynamic Audio Power Management - -platform.txt: Platform audio DMA and DAI. - -machine.txt: Machine driver internals. - -pop_clicks.txt: How to minimise audio artifacts. - -clocking.txt: ASoC clocking for best power performance. - -jack.txt: ASoC jack detection. - -DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/soc/platform.rst index 3a08a2c9150c..d5574904d981 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/soc/platform.rst @@ -1,3 +1,4 @@ +==================== ASoC Platform Driver ==================== @@ -9,21 +10,23 @@ Audio DMA ========= The platform DMA driver optionally supports the following ALSA operations:- +:: -/* SoC audio ops */ -struct snd_soc_ops { + /* SoC audio ops */ + struct snd_soc_ops { int (*startup)(struct snd_pcm_substream *); void (*shutdown)(struct snd_pcm_substream *); int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); int (*trigger)(struct snd_pcm_substream *, int); -}; + }; The platform driver exports its DMA functionality via struct snd_soc_platform_driver:- +:: -struct snd_soc_platform_driver { + struct snd_soc_platform_driver { char *name; int (*probe)(struct platform_device *pdev); @@ -44,7 +47,7 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *pcm_ops; -}; + }; Please refer to the ALSA driver documentation for details of audio DMA. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ @@ -57,11 +60,11 @@ 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) +1. Digital audio interface (DAI) description +2. Digital audio interface configuration +3. PCM's description +4. SYSCLK configuration +5. Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. @@ -71,9 +74,9 @@ 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. +1. DAPM graph +2. Mixer controls +3. DMA IO to/from DSP buffers (if applicable) +4. Definition of DSP front end (FE) PCM devices. Please see DPCM.txt for a description of item 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/soc/pops-clicks.rst index e1e74daa4497..de7eb2a6604a 100644 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ b/Documentation/sound/soc/pops-clicks.rst @@ -1,3 +1,4 @@ +===================== Audio Pops and Clicks ===================== @@ -20,10 +21,11 @@ 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 + 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. @@ -36,10 +38,11 @@ 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 + Startup Order - Input PGA --> Mixers --> ADC + + Shutdown Order - ADC --> Mixers --> Input PGA Zipper Noise |