From 849369d6c66d3054688672f97d31fceb8e8230fb Mon Sep 17 00:00:00 2001 From: root Date: Fri, 25 Dec 2015 04:40:36 +0000 Subject: initial_commit --- Documentation/sound/alsa/ALSA-Configuration.txt | 2307 +++++++++++++++++++++++ Documentation/sound/alsa/Audigy-mixer.txt | 345 ++++ Documentation/sound/alsa/Audiophile-Usb.txt | 442 +++++ Documentation/sound/alsa/Bt87x.txt | 78 + Documentation/sound/alsa/CMIPCI.txt | 254 +++ Documentation/sound/alsa/ControlNames.txt | 85 + Documentation/sound/alsa/HD-Audio-Models.txt | 428 +++++ Documentation/sound/alsa/HD-Audio.txt | 723 +++++++ Documentation/sound/alsa/Joystick.txt | 86 + Documentation/sound/alsa/MIXART.txt | 100 + Documentation/sound/alsa/OSS-Emulation.txt | 305 +++ Documentation/sound/alsa/Procfile.txt | 243 +++ Documentation/sound/alsa/README.maya44 | 163 ++ Documentation/sound/alsa/SB-Live-mixer.txt | 356 ++++ Documentation/sound/alsa/VIA82xx-mixer.txt | 8 + Documentation/sound/alsa/alsa-parameters.txt | 135 ++ Documentation/sound/alsa/emu10k1-jack.txt | 74 + Documentation/sound/alsa/hda_codec.txt | 322 ++++ Documentation/sound/alsa/hdspm.txt | 362 ++++ Documentation/sound/alsa/powersave.txt | 41 + Documentation/sound/alsa/seq_oss.html | 409 ++++ Documentation/sound/alsa/serial-u16550.txt | 88 + Documentation/sound/alsa/soc/DAI.txt | 56 + Documentation/sound/alsa/soc/clocking.txt | 51 + Documentation/sound/alsa/soc/codec.txt | 195 ++ Documentation/sound/alsa/soc/dapm.txt | 294 +++ Documentation/sound/alsa/soc/jack.txt | 71 + Documentation/sound/alsa/soc/machine.txt | 93 + Documentation/sound/alsa/soc/overview.txt | 86 + Documentation/sound/alsa/soc/platform.txt | 66 + Documentation/sound/alsa/soc/pops_clicks.txt | 52 + Documentation/sound/oss/ALS | 66 + Documentation/sound/oss/AudioExcelDSP16 | 101 + Documentation/sound/oss/CMI8330 | 153 ++ Documentation/sound/oss/ESS | 34 + Documentation/sound/oss/ESS1868 | 55 + Documentation/sound/oss/Introduction | 459 +++++ Documentation/sound/oss/MultiSound | 1137 +++++++++++ Documentation/sound/oss/OPL3 | 6 + Documentation/sound/oss/Opti | 222 +++ Documentation/sound/oss/PAS16 | 163 ++ Documentation/sound/oss/PSS | 41 + Documentation/sound/oss/PSS-updates | 88 + Documentation/sound/oss/README.OSS | 1455 ++++++++++++++ Documentation/sound/oss/README.modules | 106 ++ Documentation/sound/oss/README.ymfsb | 107 ++ Documentation/sound/oss/SoundPro | 105 ++ Documentation/sound/oss/Soundblaster | 53 + Documentation/sound/oss/Tropez+ | 26 + Documentation/sound/oss/VIBRA16 | 80 + Documentation/sound/oss/WaveArtist | 170 ++ Documentation/sound/oss/btaudio | 92 + Documentation/sound/oss/mwave | 185 ++ Documentation/sound/oss/oss-parameters.txt | 51 + Documentation/sound/oss/ultrasound | 30 + Documentation/sound/oss/vwsnd | 293 +++ 56 files changed, 13596 insertions(+) create mode 100644 Documentation/sound/alsa/ALSA-Configuration.txt create mode 100644 Documentation/sound/alsa/Audigy-mixer.txt create mode 100644 Documentation/sound/alsa/Audiophile-Usb.txt create mode 100644 Documentation/sound/alsa/Bt87x.txt create mode 100644 Documentation/sound/alsa/CMIPCI.txt create mode 100644 Documentation/sound/alsa/ControlNames.txt create mode 100644 Documentation/sound/alsa/HD-Audio-Models.txt create mode 100644 Documentation/sound/alsa/HD-Audio.txt create mode 100644 Documentation/sound/alsa/Joystick.txt create mode 100644 Documentation/sound/alsa/MIXART.txt create mode 100644 Documentation/sound/alsa/OSS-Emulation.txt create mode 100644 Documentation/sound/alsa/Procfile.txt create mode 100644 Documentation/sound/alsa/README.maya44 create mode 100644 Documentation/sound/alsa/SB-Live-mixer.txt create mode 100644 Documentation/sound/alsa/VIA82xx-mixer.txt create mode 100644 Documentation/sound/alsa/alsa-parameters.txt create mode 100644 Documentation/sound/alsa/emu10k1-jack.txt create mode 100644 Documentation/sound/alsa/hda_codec.txt create mode 100644 Documentation/sound/alsa/hdspm.txt create mode 100644 Documentation/sound/alsa/powersave.txt create mode 100644 Documentation/sound/alsa/seq_oss.html create mode 100644 Documentation/sound/alsa/serial-u16550.txt create mode 100644 Documentation/sound/alsa/soc/DAI.txt create mode 100644 Documentation/sound/alsa/soc/clocking.txt create mode 100644 Documentation/sound/alsa/soc/codec.txt create mode 100644 Documentation/sound/alsa/soc/dapm.txt create mode 100644 Documentation/sound/alsa/soc/jack.txt create mode 100644 Documentation/sound/alsa/soc/machine.txt create mode 100644 Documentation/sound/alsa/soc/overview.txt create mode 100644 Documentation/sound/alsa/soc/platform.txt create mode 100644 Documentation/sound/alsa/soc/pops_clicks.txt create mode 100644 Documentation/sound/oss/ALS create mode 100644 Documentation/sound/oss/AudioExcelDSP16 create mode 100644 Documentation/sound/oss/CMI8330 create mode 100644 Documentation/sound/oss/ESS create mode 100644 Documentation/sound/oss/ESS1868 create mode 100644 Documentation/sound/oss/Introduction create mode 100644 Documentation/sound/oss/MultiSound create mode 100644 Documentation/sound/oss/OPL3 create mode 100644 Documentation/sound/oss/Opti create mode 100644 Documentation/sound/oss/PAS16 create mode 100644 Documentation/sound/oss/PSS create mode 100644 Documentation/sound/oss/PSS-updates create mode 100644 Documentation/sound/oss/README.OSS create mode 100644 Documentation/sound/oss/README.modules create mode 100644 Documentation/sound/oss/README.ymfsb create mode 100644 Documentation/sound/oss/SoundPro create mode 100644 Documentation/sound/oss/Soundblaster create mode 100644 Documentation/sound/oss/Tropez+ create mode 100644 Documentation/sound/oss/VIBRA16 create mode 100644 Documentation/sound/oss/WaveArtist create mode 100644 Documentation/sound/oss/btaudio create mode 100644 Documentation/sound/oss/mwave create mode 100644 Documentation/sound/oss/oss-parameters.txt create mode 100644 Documentation/sound/oss/ultrasound create mode 100644 Documentation/sound/oss/vwsnd (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt new file mode 100644 index 00000000..89757012 --- /dev/null +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -0,0 +1,2307 @@ + + 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 resouces. 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 (0 = auto, 1 = use LPIB, 2 = POSBUF) + 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, 2= + dynamic registration via mute switch on/off); the 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) + + 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=" 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 "genric" 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.) + + 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 hardwares, 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 hardwares 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 + * 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 endianess. + + 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) + async_unlink - Use async unlink mode (default: yes) + device_setup - Device specific magic number (optional) + - Influence depends on the device + - Default: 0x0000 + ignore_ctl_error - Ignore any USB-controller regarding mixer + interface (default: no) + + This module supports multiple devices, autoprobe and hotplugging. + + NB: nrpacks parameter can be modified dynamically via sysfs. + Don't put the value over 20. Changing via sysfs has no sanity + check. + NB: async_unlink=0 would cause Oops. It remains just for + debugging purpose (if any). + NB: ignore_ctl_error=1 may help when you get an error at accessing + the mixer element such as URB error -22. This happens on some + buggy USB device or the controller. + + Module snd-usb-caiaq + -------------------- + + Module for caiaq UB audio interfaces, + * Native Instruments RigKontrol2 + * Native Instruments Kore Controller + * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 8 DJ + + This module supports multiple devices, autoprobe and hotplugging. + + Module snd-usb-usx2y + -------------------- + + Module for Tascam USB US-122, US-224 and US-428 devices. + + This module supports multiple devices, autoprobe and hotplugging. + + Note: you need to load the firmware via usx2yloader utility included + in alsa-tools and alsa-firmware packages. + + Module snd-via82xx + ------------------ + + Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, + 8233A, 8233C, 8235, 8237 (south) bridge. + + mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup + [VIA686A/686B only] + joystick - Enable joystick (default off) [VIA686A/686B only] + ac97_clock - AC'97 codec clock base (default 48000Hz) + dxs_support - support DXS channels, + 0 = auto (default), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA, 5 = enable any sample + rate and different sample rates on different + channels + [VIA8233/C, 8235, 8237 only] + ac97_quirk - AC'97 workaround for strange hardware + See "AC97 Quirk Option" section below. + + This module supports one chip and autoprobe. + + Note: on some SMP motherboards like MSI 694D the interrupts might + not be generated properly. In such a case, please try to + set the SMP (or MPS) version on BIOS to 1.1 instead of + default value 1.4. Then the interrupt number will be + assigned under 15. You might also upgrade your BIOS. + + Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) + channels as the first PCM. On these channels, up to 4 + streams can be played at the same time, and the controller + can perform sample rate conversion with separate rates for + each channel. + As default (dxs_support = 0), 48k fixed rate is chosen + except for the known devices since the output is often + noisy except for 48k on some mother boards due to the + bug of BIOS. + Please try once dxs_support=5 and if it works on other + sample rates (e.g. 44.1kHz of mp3 playback), please let us + know the PCI subsystem vendor/device id's (output of + "lspci -nv"). + If dxs_support=5 does not work, try dxs_support=4; if it + doesn't work too, try dxs_support=1. (dxs_support=1 is + usually for old motherboards. The correct implemented + board should work with 4 or 5.) If it still doesn't + work and the default setting is ok, dxs_support=3 is the + right choice. If the default setting doesn't work at all, + try dxs_support=2 to disable the DXS channels. + In any cases, please let us know the result and the + subsystem vendor/device ids. See "Links and Addresses" + below. + + Note: for the MPU401 on VIA823x, use snd-mpu401 driver + additionally. The mpu_port option is for VIA686 chips only. + + The power-management is supported. + + Module snd-via82xx-modem + ------------------------ + + Module for VIA82xx AC97 modem + + ac97_clock - AC'97 codec clock base (default 48000Hz) + + This module supports one card and autoprobe. + + Note: The default index value of this module is -2, i.e. the first + slot is excluded. + + The power-management is supported. + + Module snd-virmidi + ------------------ + + Module for virtual rawmidi devices. + This module creates virtual rawmidi devices which communicate + to the corresponding ALSA sequencer ports. + + midi_devs - MIDI devices # (1-4, default=4) + + This module supports multiple cards. + + Module snd-virtuoso + ------------------- + + Module for sound cards based on the Asus AV66/AV100/AV200 chips, + i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX, + 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.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.conf +alias snd-card-0 snd-emu10k1 +alias sound-slot-0 snd-emu10k1 +----- /etc/modprobe.conf + +The available number of auto-loaded sound cards depends on the module +option "cards_limit" of snd module. As default it's set to 1. +To enable the auto-loading of multiple cards, specify the number of +sound cards in that option. + +When multiple cards are available, it'd better to specify the index +number for each card via module option, too, so that the order of +cards is kept consistent. + +An example configuration for two sound cards is like below: + +----- /etc/modprobe.conf +# ALSA portion +options snd cards_limit=2 +alias snd-card-0 snd-interwave +alias snd-card-1 snd-ens1371 +options snd-interwave index=0 +options snd-ens1371 index=1 +# OSS/Free portion +alias sound-slot-0 snd-interwave +alias sound-slot-1 snd-ens1371 +----- /etc/modprobe.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 " []" + + - name of application with (higher priority) or without path + - number of fragments or zero if auto + - size of fragment in bytes or zero if auto + - optional parameters + - disable the application tries to open a pcm device for + this channel but does not want to use it. + (Cause a bug or mmap needs) + It's good for Quake etc... + - direct don't use plugins + - block force block mode (rvplayer) + - non-block force non-block mode + - whole-frag write only whole fragments (optimization affecting + playback only) + - no-silence do not fill silence ahead to avoid clicks + - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl + instead of filled blocks + + Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss + echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss + echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss + + +Early Buffer Allocation +======================= + +Some drivers (e.g. hdsp) require the large contiguous buffers, and +sometimes it's too late to find such spaces when the driver module is +actually loaded due to memory fragmentation. You can pre-allocate the +PCM buffers by loading snd-page-alloc module and write commands to its +proc file in prior, for example, in the early boot stage like +/etc/init.d/*.local scripts. + +Reading the proc file /proc/drivers/snd-page-alloc shows the current +usage of page allocation. In writing, you can send the following +commands to the snd-page-alloc driver: + + - add VENDOR DEVICE MASK SIZE BUFFERS + + VENDOR and DEVICE are PCI vendor and device IDs. They take + integer numbers (0x prefix is needed for the hex). + MASK is the PCI DMA mask. Pass 0 if not restricted. + SIZE is the size of each buffer to allocate. You can pass + k and m suffix for KB and MB. The max number is 16MB. + BUFFERS is the number of buffers to allocate. It must be greater + than 0. The max number is 4. + + - erase + + This will erase the all pre-allocated buffers which are not in + use. + + +Links and Addresses +=================== + + ALSA project homepage + http://www.alsa-project.org + + Kernel Bugzilla + http://bugzilla.kernel.org/ + + ALSA Developers ML + mailto:alsa-devel@alsa-project.org + + alsa-info.sh script + http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt new file mode 100644 index 00000000..7f10dc6f --- /dev/null +++ b/Documentation/sound/alsa/Audigy-mixer.txt @@ -0,0 +1,345 @@ + + Sound Blaster Audigy mixer / default DSP code + =========================================== + +This is based on SB-Live-mixer.txt. + +The EMU10K2 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K2 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + +name='PCM Front Playback Volume',index=0 + +This control is used to attenuate samples for left and right front PCM FX-bus +accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM +samples for 5.1 playback. The result samples are forwarded to the front DAC PCM +slots of the Philips DAC. + +name='PCM Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right surround PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM +samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM +slots of the Philips DAC. + +name='PCM Center Playback Volume',index=0 + +This control is used to attenuate samples for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample +is forwarded to the center DAC PCM slot of the Philips DAC. + +name='PCM LFE Playback Volume',index=0 + +This control is used to attenuate sample for LFE PCM FX-bus accumulator. +ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample +is forwarded to the LFE DAC PCM slot of the Philips DAC. + +name='PCM Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for +stereo playback. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='PCM Capture Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Music Playback Volume',index=0 + +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Music Capture Volume',index=0 + +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Mic Playback Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +For Mic input is used AC97 codec. The result samples are forwarded to +the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic +capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. + +name='Mic Capture Volume',index=0 + +This control is used to attenuate samples for left and right Mic input. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Audigy CD Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the Philips DAC. + +name='Audigy CD Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 Optical Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the Philips DAC. + +name='IEC958 Optical Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Line2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Analog Mix Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs from Philips ADC. The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. This contains mix from analog sources +like CD, Line In, Aux, .... + +name='Analog Mix Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs Philips ADC. The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Aux2 Playback Volume',index=0 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the front +DAC PCM slots of the Philips DAC. + +name='Aux2 Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the AudigyDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Front Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right front speakers of +this mix. + +name='Surround Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate samples for left and right surround speakers of +this mix. + +name='Center Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for center speaker of this mix. + +name='LFE Playback Volume',index=0 + +All stereo signals are mixed together and mirrored to surround, center and LFE. +This control is used to attenuate sample for LFE speaker of this mix. + +name='Tone Control - Switch',index=0 + +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 + +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 + +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Master Playback Volume',index=0 + +This control is used to attenuate samples for front, surround, center and +LFE outputs. + +name='IEC958 Optical Raw Playback Switch',index=0 + +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + + +2) PCM stream related controls +------------------------------ + +name='EMU10K1 PCM Volume',index 0-31 + +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + + 0 - mono, default 0xffff (no attenuation) + 1 - left, default 0xffff (no attenuation) + 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 + +This control specifies the destination - FX-bus accumulators. There 24 +values with this mapping: + + 0 - mono, A destination (FX-bus 0-63), default 0 + 1 - mono, B destination (FX-bus 0-63), default 1 + 2 - mono, C destination (FX-bus 0-63), default 2 + 3 - mono, D destination (FX-bus 0-63), default 3 + 4 - mono, E destination (FX-bus 0-63), default 0 + 5 - mono, F destination (FX-bus 0-63), default 0 + 6 - mono, G destination (FX-bus 0-63), default 0 + 7 - mono, H destination (FX-bus 0-63), default 0 + 8 - left, A destination (FX-bus 0-63), default 0 + 9 - left, B destination (FX-bus 0-63), default 1 + 10 - left, C destination (FX-bus 0-63), default 2 + 11 - left, D destination (FX-bus 0-63), default 3 + 12 - left, E destination (FX-bus 0-63), default 0 + 13 - left, F destination (FX-bus 0-63), default 0 + 14 - left, G destination (FX-bus 0-63), default 0 + 15 - left, H destination (FX-bus 0-63), default 0 + 16 - right, A destination (FX-bus 0-63), default 0 + 17 - right, B destination (FX-bus 0-63), default 1 + 18 - right, C destination (FX-bus 0-63), default 2 + 19 - right, D destination (FX-bus 0-63), default 3 + 20 - right, E destination (FX-bus 0-63), default 0 + 21 - right, F destination (FX-bus 0-63), default 0 + 22 - right, G destination (FX-bus 0-63), default 0 + 23 - right, H destination (FX-bus 0-63), default 0 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 + +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + + 0 - mono, A destination attn, default 255 (no attenuation) + 1 - mono, B destination attn, default 255 (no attenuation) + 2 - mono, C destination attn, default 0 (mute) + 3 - mono, D destination attn, default 0 (mute) + 4 - mono, E destination attn, default 0 (mute) + 5 - mono, F destination attn, default 0 (mute) + 6 - mono, G destination attn, default 0 (mute) + 7 - mono, H destination attn, default 0 (mute) + 8 - left, A destination attn, default 255 (no attenuation) + 9 - left, B destination attn, default 0 (mute) + 10 - left, C destination attn, default 0 (mute) + 11 - left, D destination attn, default 0 (mute) + 12 - left, E destination attn, default 0 (mute) + 13 - left, F destination attn, default 0 (mute) + 14 - left, G destination attn, default 0 (mute) + 15 - left, H destination attn, default 0 (mute) + 16 - right, A destination attn, default 0 (mute) + 17 - right, B destination attn, default 255 (no attenuation) + 18 - right, C destination attn, default 0 (mute) + 19 - right, D destination attn, default 0 (mute) + 20 - right, E destination attn, default 0 (mute) + 21 - right, F destination attn, default 0 (mute) + 22 - right, G destination attn, default 0 (mute) + 23 - right, H destination attn, default 0 (mute) + + + +4) MANUALS/PATENTS: +------------------- + +ftp://opensource.creative.com/pub/doc +------------------------------------- + + Files: + LM4545.pdf AC97 Codec + + m2049.pdf The EMU10K1 Digital Audio Processor + + hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + Patent numbers: + WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) + streams + + WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + + WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + + US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + + US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + + US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + + US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) + + US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + + US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) + pool of short term memory registers + + US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 00000000..a4c53d89 --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,442 @@ + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 + ======================================================== + + Thibault Le Meur + +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 + - Modifying document structure +* v1.5 - Thibault Le Meur (2007-07-12) + - Added AC3/DTS passthru info + + +1 - Audiophile USB Specs and correct usage +========================================== + +This part is a reminder of important facts about the functions and limitations +of the device. + +The device has 4 audio interfaces, and 2 MIDI ports: + * Analog Stereo Input (Ai) + - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) + - When the 1/4" TS (jack) connectors are connected, the RCA connectors + are disabled + * Analog Stereo Output (Ao) + * Digital Stereo Input (Di) + * Digital Stereo Output (Do) + * Midi In (Mi) + * Midi Out (Mo) + +The internal DAC/ADC has the following characteristics: +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two interfaces can't use different sample depths at the same time. +Moreover, the Audiophile USB documentation gives the following Warning: +"Please exit any audio application running before switching between bit depths" + +Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be +activated at the same time depending on the audio mode selected: + * 16-bit/48kHz ==> 4 channels in + 4 channels out + - Ai+Ao+Di+Do + * 24-bit/48kHz ==> 4 channels in + 2 channels out, + or 2 channels in + 4 channels out + - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do + * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) + - Ai or Ao or Di or Do + +Important facts about the Digital interface: +-------------------------------------------- + * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under Linux + - Note that in this setup only the Do interface can be enabled + * Apart from recording an audio digital stream, enabling the Di port is a way +to synchronize the device to an external sample clock + - As a consequence, the Di port must be enable only if an active Digital +source is connected + - Enabling Di when no digital source is connected can result in a +synchronization error (for instance sound played at an odd sample rate) + + +2 - Audiophile USB MIDI support in ALSA +======================================= + +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + * snd-usb-audio + * snd-seq-midi + +No additional setting is required. + + +3 - Audiophile USB Audio support in ALSA +======================================== + +Audio functions of the Audiophile USB device are handled by the snd-usb-audio +module. This module can work in a default mode (without any device-specific +parameter), or in an "advanced" mode with the device-specific parameter called +"device_setup". + +3.1 - Default Alsa driver mode +------------------------------ + +The default behavior of the snd-usb-audio driver is to list the device +capabilities at startup and activate the required mode when required +by the applications: for instance if the user is recording in a +24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, +the snd-usb-audio module will reconfigure the device on the fly. + +This approach has the advantage to let the driver automatically switch from sample +rates/depths automatically according to the user's needs. However, those who +are using the device under windows know that this is not how the device is meant to +work: under windows applications must be closed before using the m-audio control +panel to switch the device working mode. Thus as we'll see in next section, this +Default Alsa driver mode can lead to device misconfigurations. + +Let's get back to the Default Alsa driver mode for now. In this case the +Audiophile interfaces are mapped to alsa pcm devices in the following +way (I suppose the device's index is 1): + * hw:1,0 is Ao in playback and Di in capture + * hw:1,1 is Do in playback and Ai in capture + * hw:1,2 is Do in AC3/DTS passthrough mode + +In this mode, the device uses Big Endian byte-encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. + +One exception is the hw:1,2 port which was reported to be Little Endian +compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. +This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface +is reported to be big endian in this default driver mode. + +Examples: + * playing a S24_3BE encoded raw file to the Ao port + % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw + * recording a S24_3BE encoded raw file from the Ai port + % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw + * playing a S16_BE encoded raw file to the Do port + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + * playing an ac3 sample file to the Do port + % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw + +If you're happy with the default Alsa driver mode and don't experience any +issue with this mode, then you can skip the following chapter. + +3.2 - Advanced module setup +--------------------------- + +Due to the hardware constraints described above, the device initialization made +by the Alsa driver in default mode may result in a corrupted state of the +device. For instance, a particularly annoying issue is that the sound captured +from the Ai interface sounds distorted (as if boosted with an excessive high +volume gain). + +For people having this problem, the snd-usb-audio module has a new module +parameter called "device_setup" (this parameter was introduced in kernel +release 2.6.17) + +3.2.1 - Initializing the working mode of the Audiophile USB + +As far as the Audiophile USB device is concerned, this value let the user +specify: + * the sample depth + * the sample rate + * whether the Di port is used or not + +When initialized with "device_setup=0x00", the snd-usb-audio module has +the same behaviour as when the parameter is omitted (see paragraph "Default +Alsa driver mode" above) + +Others modes are described in the following subsections. + +3.2.1.1 - 16-bit modes + +The two supported modes are: + + * device_setup=0x01 + - 16bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x11 + - 16bits 48kHz mode with Di enabled + - Ai,Ao,Di,Do can be used at the same time + - hw:1,0 is available in capture mode + - hw:1,2 is not available + +In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, +the devices where reported to be Big-Endian when in fact they were Little-Endian +so that playing a file was a matter of using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw +where "test_S16_LE.raw" was in fact a little-endian sample file. + +Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in +these modes) a fix has been committed (expected in kernel 2.6.23) and +Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as +using: + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw + +3.2.1.2 - 24-bit modes + +The three supported modes are: + + * device_setup=0x09 + - 24bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + + * device_setup=0x19 + - 24bits 48kHz mode with Di enabled + - 3 ports from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in capture mode and an active digital source must be + connected to Di + - hw:1,2 is not available + + * device_setup=0x0D or 0x10 + - 24bits 96kHz mode + - Di is enabled by default for this mode but does not need to be connected + to an active source + - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in captured mode + - hw:1,2 is not available + +In these modes the device is only Big-Endian compliant (see "Default Alsa driver +mode" above for an aplay command example) + +3.2.1.3 - AC3 w/ DTS passthru mode + +Thanks to Hakan Lennestal, I now have a report saying that this mode works. + + * device_setup=0x03 + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +The command line used to playback the AC3/DTS encoded .wav-files in this mode: + % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw + +3.2.2 - How to use the device_setup parameter +---------------------------------------------- + +The parameter can be given: + + * By manually probing the device (as root): + # modprobe -r snd-usb-audio + # modprobe snd-usb-audio index=1 device_setup=0x09 + + * Or while configuring the modules options in your modules configuration file + - For Fedora distributions, edit the /etc/modprobe.conf file: + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +CAUTION when initializing the device +------------------------------------- + + * Correct initialization on the device requires that device_setup is given to + the module BEFORE the device is turned on. So, if you use the "manual probing" + method described above, take care to power-on the device AFTER this initialization. + + * Failing to respect this will lead to a misconfiguration of the device. In this case + turn off the device, unprobe the snd-usb-audio module, then probe it again with + correct device_setup parameter and then (and only then) turn on the device again. + + * If you've correctly initialized the device in a valid mode and then want to switch + to another mode (possibly with another sample-depth), please use also the following + procedure: + - first turn off the device + - de-register the snd-usb-audio module (modprobe -r) + - change the device_setup parameter by changing the device_setup + option in /etc/modprobe.conf + - turn on the device + * A workaround for this last issue has been applied to kernel 2.6.23, but it may not + be enough to ensure the 'stability' of the device initialization. + +3.2.3 - Technical details for hackers +------------------------------------- +This section is for hackers, wanting to understand details about the device +internals and how Alsa supports it. + +3.2.3.1 - Audiophile USB's device_setup structure + +If you want to understand the device_setup magic numbers for the Audiophile +USB, you need some very basic understanding of binary computation. However, +this is not required to use the parameter and you may skip this section. + +The device_setup is one byte long and its structure is the following: + + +---+---+---+---+---+---+---+---+ + | b7| b6| b5| b4| b3| b2| b1| b0| + +---+---+---+---+---+---+---+---+ + | 0 | 0 | 0 | Di|24B|96K|DTS|SET| + +---+---+---+---+---+---+---+---+ + +Where: + * b0 is the "SET" bit + - it MUST be set if device_setup is initialized + * b1 is the "DTS" bit + - it is set only for Digital output with DTS/AC3 + - this setup is not tested + * b2 is the Rate selection flag + - When set to "1" the rate range is 48.1-96kHz + - Otherwise the sample rate range is 8-48kHz + * b3 is the bit depth selection flag + - When set to "1" samples are 24bits long + - Otherwise they are 16bits long + - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits + samples + * b4 is the Digital input flag + - When set to "1" the device assumes that an active digital source is + connected + - You shouldn't enable Di if no source is seen on the port (this leads to + synchronization issues) + - b4 is implied by b2 (since only one port is enabled at a time no synch + error can occur) + * b5 to b7 are reserved for future uses, and must be set to "0" + - might become Ao, Do, Ai, for b7, b6, b4 respectively + +Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since + b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll + only be able to use one at the same time + +3.2.3.2 - USB implementation details for this device + +You may safely skip this section if you're not interested in driver +hacking. + +This section describes some internal aspects of the device and summarizes the +data I got by usb-snooping the windows and Linux drivers. + +The M-Audio Audiophile USB has 7 USB Interfaces: +a "USB interface": + * USB Interface nb.0 + * USB Interface nb.1 + - Audio Control function + * USB Interface nb.2 + - Analog Output + * USB Interface nb.3 + - Digital Output + * USB Interface nb.4 + - Analog Input + * USB Interface nb.5 + - Digital Input + * USB Interface nb.6 + - MIDI interface compliant with the MIDIMAN quirk + +Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: + * Interface 3 (Digital Out) has an extra Alset nb.6 + * Interface 5 (Digital In) does not have Alset nb.3 and 5 + +Here is a short description of the AltSettings capabilities: + * AltSettings 1 corresponds to + - 24-bit depth, 48.1-96kHz sample mode + - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) + * AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 6 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch playback (Do), audio format type III IEC1937_AC-3 + +In order to ensure a correct initialization of the device, the driver +_must_know_ how the device will be used: + * if DTS is chosen, only Interface 2 with AltSet nb.6 must be + registered + * if 96KHz only AltSets nb.1 of each interface must be selected + * if samples are using 24bits/48KHz then AltSet 2 must me used if + Digital input is connected, and only AltSet nb.3 if Digital input + is not connected + * if samples are using 16bits/48KHz then AltSet 4 must me used if + Digital input is connected, and only AltSet nb.5 if Digital input + is not connected + +When device_setup is given as a parameter to the snd-usb-audio module, the +parse_audio_endpoints function uses a quirk called +"audiophile_skip_setting_quirk" in order to prevent AltSettings not +corresponding to device_setup from being registered in the driver. + +4 - Audiophile USB and Jack support +=================================== + +This section deals with support of the Audiophile USB device in Jack. + +There are 2 main potential issues when using Jackd with the device: +* support for Big-Endian devices in 24-bit modes +* support for 4-in / 4-out channels + +4.1 - Direct support in Jackd +----------------------------- + +Jack supports big endian devices only in recent versions (thanks to +Andreas Steinmetz for his first big-endian patch). I can't remember +exactly when this support was released into jackd, let's just say that +with jackd version 0.103.0 it's almost ok (just a small bug is affecting +16bits Big-Endian devices, but since you've read carefully the above +paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices +are now Little Endians ;-) ). + +You can run jackd with the following command for playback with Ao and +record with Ai: + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + +4.2 - Using Alsa plughw +----------------------- +If you don't have a recent Jackd installed, you can downgrade to using +the Alsa "plug" converter. + +For instance here is one way to run Jack with 2 playback channels on Ao and 2 +capture channels from Ai: + % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 + +However you may see the following warning message: +"You appear to be using the ALSA software "plug" layer, probably a result of +using the "default" ALSA device. This is less efficient than it could be. +Consider using a hardware device instead rather than using the plug layer." + +4.3 - Getting 2 input and/or output interfaces in Jack +------------------------------------------------------ + +As you can see, starting the Jack server this way will only enable 1 stereo +input (Di or Ai) and 1 stereo output (Ao or Do). + +This is due to the following restrictions: +* Jack can only open one capture device and one playback device at a time +* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 + (and optionally hw:1,2) + +If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to +combine the Alsa devices into one logical "complex" device. + +If you want to give it a try, I recommend reading the information from +this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html +It is related to another device (ice1712) but can be adapted to suit +the Audiophile USB. + +Enabling multiple Audiophile USB interfaces for Jackd will certainly require: +* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) +* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) +* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc + file +* start jackd with this device + +I had no success in testing this for now, if you have any success with this kind +of setup, please drop me an email. diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt new file mode 100644 index 00000000..f158cde8 --- /dev/null +++ b/Documentation/sound/alsa/Bt87x.txt @@ -0,0 +1,78 @@ +Intro +===== + +You might have noticed that the bt878 grabber cards have actually +_two_ PCI functions: + +$ lspci +[ ... ] +00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) +00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02) +[ ... ] + +The first does video, it is backward compatible to the bt848. The second +does audio. snd-bt87x is a driver for the second function. It's a sound +driver which can be used for recording sound (and _only_ recording, no +playback). As most TV cards come with a short cable which can be plugged +into your sound card's line-in you probably don't need this driver if all +you want to do is just watching TV... + +Some cards do not bother to connect anything to the audio input pins of +the chip, and some other cards use the audio function to transport MPEG +video data, so it's quite possible that audio recording may not work +with your card. + + +Driver Status +============= + +The driver is now stable. However, it doesn't know about many TV cards, +and it refuses to load for cards it doesn't know. + +If the driver complains ("Unknown TV card found, the audio driver will +not load"), you can specify the load_all=1 option to force the driver to +try to use the audio capture function of your card. If the frequency of +recorded data is not right, try to specify the digital_rate option with +other values than the default 32000 (often it's 44100 or 64000). + +If you have an unknown card, please mail the ID and board name to +, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. + + +Audio modes +=========== + +The chip knows two different modes (digital/analog). snd-bt87x +registers two PCM devices, one for each mode. They cannot be used at +the same time. + + +Digital audio mode +================== + +The first device (hw:X,0) gives you 16 bit stereo sound. The sample +rate depends on the external source which feeds the Bt87x with digital +sound via I2S interface. + + +Analog audio mode (A/D) +======================= + +The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported +sample rates are between 119466 and 448000 Hz (yes, these numbers are +that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the +maximum sample rate is 1792000 Hz, but audio data becomes unusable +beyond 896000 Hz on my card. + +The chip has three analog inputs. Consequently you'll get a mixer +device to control these. + + +Have fun, + + Clemens + + +Written by Clemens Ladisch +big parts copied from btaudio.txt by Gerd Knorr diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt new file mode 100644 index 00000000..16935c85 --- /dev/null +++ b/Documentation/sound/alsa/CMIPCI.txt @@ -0,0 +1,254 @@ + Brief Notes on C-Media 8338/8738/8768/8770 Driver + ================================================= + + Takashi Iwai + + +Front/Rear Multi-channel Playback +--------------------------------- + +CM8x38 chip can use ADC as the second DAC so that two different stereo +channels can be used for front/rear playbacks. Since there are two +DACs, both streams are handled independently unlike the 4/6ch multi- +channel playbacks in the section below. + +As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for +card#0) for front and 4/6ch playbacks, while the second PCM device +(hw:0,1) is assigned to the second DAC for rear playback. + +There are slight differences between the two DACs: + +- The first DAC supports U8 and S16LE formats, while the second DAC + supports only S16LE. +- The second DAC supports only two channel stereo. + +Please note that the CM8x38 DAC doesn't support continuous playback +rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000, +44100 and 48000 Hz. + +The rear output can be heard only when "Four Channel Mode" switch is +disabled. Otherwise no signal will be routed to the rear speakers. +As default it's turned on. + +*** WARNING *** +When "Four Channel Mode" switch is off, the output from rear speakers +will be FULL VOLUME regardless of Master and PCM volumes. +This might damage your audio equipment. Please disconnect speakers +before your turn off this switch. +*** WARNING *** + +[ Well.. I once got the output with correct volume (i.e. same with the + front one) and was so excited. It was even with "Four Channel" bit + on and "double DAC" mode. Actually I could hear separate 4 channels + from front and rear speakers! But.. after reboot, all was gone. + It's a very pity that I didn't save the register dump at that + time.. Maybe there is an unknown register to achieve this... ] + +If your card has an extra output jack for the rear output, the rear +playback should be routed there as default. If not, there is a +control switch in the driver "Line-In As Rear", which you can change +via alsamixer or somewhat else. When this switch is on, line-in jack +is used as rear output. + +There are two more controls regarding to the rear output. +The "Exchange DAC" switch is used to exchange front and rear playback +routes, i.e. the 2nd DAC is output from front output. + + +4/6 Multi-Channel Playback +-------------------------- + +The recent CM8738 chips support for the 4/6 multi-channel playback +function. This is useful especially for AC3 decoding. + +When the multi-channel is supported, the driver name has a suffix +"-MC" such like "CMI8738-MC6". You can check this name from +/proc/asound/cards. + +When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or +4) channels. While the dual DAC supports two different rates or +formats, the 4/6-ch playback supports only the same condition for all +channels. Since the multi-channel playback mode uses both DACs, you +cannot operate with full-duplex. + +The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" +in alsa-lib. For example, you can play a WAV file with 6 channels like + + % aplay -Dsurround51 sixchannels.wav + +For programming the 4/6 channel playback, you need to specify the PCM +channels as you like and set the format S16LE. For example, for playback +with 4 channels, + + snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); + // or mmap if you like + snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE); + snd_pcm_hw_params_set_channels(pcm, hw, 4); + +and use the interleaved 4 channel data. + +There are some control switchs affecting to the speaker connections: + +"Line-In Mode" - an enum control to change the behavior of line-in + jack. Either "Line-In", "Rear Output" or "Bass Output" can + be selected. The last item is available only with model 039 + or newer. + When "Rear Output" is chosen, the surround channels 3 and 4 + are output to line-in jack. +"Mic-In Mode" - an enum control to change the behavior of mic-in + jack. Either "Mic-In" or "Center/LFE Output" can be + selected. + When "Center/LFE Output" is chosen, the center and bass + channels (channels 5 and 6) are output to mic-in jack. + +Digital I/O +----------- + +The CM8x38 provides the excellent SPDIF capability with very cheap +price (yes, that's the reason I bought the card :) + +The SPDIF playback and capture are done via the third PCM device +(hw:0,2). Usually this is assigned to the PCM device "spdif". +The available rates are 44100 and 48000 Hz. +For playback with aplay, you can run like below: + + % aplay -Dhw:0,2 foo.wav + +or + + % aplay -Dspdif foo.wav + +24bit format is also supported experimentally. + +The playback and capture over SPDIF use normal DAC and ADC, +respectively, so you cannot playback both analog and digital streams +simultaneously. + +To enable SPDIF output, you need to turn on "IEC958 Output Switch" +control via mixer or alsactl ("IEC958" is the official name of +so-called S/PDIF). Then you'll see the red light on from the card so +you know that's working obviously :) +The SPDIF input is always enabled, so you can hear SPDIF input data +from line-out with "IEC958 In Monitor" switch at any time (see +below). + +You can play via SPDIF even with the first device (hw:0,0), +but SPDIF is enabled only when the proper format (S16LE), sample rate +(441100 or 48000) and channels (2) are used. Otherwise it's turned +off. (Also don't forget to turn on "IEC958 Output Switch", too.) + + +Additionally there are relevant control switches: + +"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and + output through SPDIF. This switch appears only on old chip + models (CM8738 033 and 037). + Note: without this control you can output PCM to SPDIF. + This is "mixing" of streams, so e.g. it's not for AC3 output + (see the next section). + +"IEC958 In Select" - Select SPDIF input, the internal CD-in (false) + and the external input (true). + +"IEC958 Loop" - SPDIF input data is loop back into SPDIF + output (aka bypass) + +"IEC958 Copyright" - Set the copyright bit. + +"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface. + On some cards this doesn't work and you need to change the + configuration with hardware dip-switch. + +"IEC958 In Monitor" - SPDIF input is routed to DAC. + +"IEC958 In Phase Inverse" - Set SPDIF input format as inverse. + [FIXME: this doesn't work on all chips..] + +"IEC958 In Valid" - Set input validity flag detection. + +Note: When "PCM Playback Switch" is on, you'll hear the digital output +stream through analog line-out. + + +The AC3 (RAW DIGITAL) OUTPUT +---------------------------- + +The driver supports raw digital (typically AC3) i/o over SPDIF. This +can be toggled via IEC958 playback control, but usually you need to +access it via alsa-lib. See alsa-lib documents for more details. + +On the raw digital mode, the "PCM Playback Switch" is automatically +turned off so that non-audio data is heard from the analog line-out. +Similarly the following switches are off: "IEC958 Mix Analog" and +"IEC958 Loop". The switches are resumed after closing the SPDIF PCM +device automatically to the previous state. + +On the model 033, AC3 is implemented by the software conversion in +the alsa-lib. If you need to bypass the software conversion of IEC958 +subframes, pass the "soft_ac3=0" module option. This doesn't matter +on the newer models. + + +ANALOG MIXER INTERFACE +---------------------- + +The mixer interface on CM8x38 is similar to SB16. +There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback +volumes. Synth, CD, Line and Mic have playback and capture switches, +too, as well as SB16. + +In addition to the standard SB mixer, CM8x38 provides more functions. +- PCM playback switch +- PCM capture switch (to capture the data sent to DAC) +- Mic Boost switch +- Mic capture volume +- Aux playback volume/switch and capture switch +- 3D control switch + + +MIDI CONTROLLER +--------------- + +With CMI8338 chips, the MPU401-UART interface is disabled as default. +You need to set the module option "mpu_port" to a valid I/O port address +to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and +0x330. Choose a value that doesn't conflict with other cards. + +With CMI8738 and newer chips, the MIDI interface is enabled by default +and the driver automatically chooses a port address. + +There is _no_ hardware wavetable function on this chip (except for +OPL3 synth below). +What's said as MIDI synth on Windows is a software synthesizer +emulation. On Linux use TiMidity or other softsynth program for +playing MIDI music. + + +FM OPL/3 Synth +-------------- + +The FM OPL/3 is also enabled as default only for the first card. +Set "fm_port" module option for more cards. + +The output quality of FM OPL/3 is, however, very weird. +I don't know why.. + +CMI8768 and newer chips do not have the FM synth. + + +Joystick and Modem +------------------ + +The legacy joystick is supported. To enable the joystick support, pass +joystick_port=1 module option. The value 1 means the auto-detection. +If the auto-detection fails, try to pass the exact I/O address. + +The modem is enabled dynamically via a card control switch "Modem". + + +Debugging Information +--------------------- + +The registers are shown in /proc/asound/cardX/cmipci. If you have any +problem (especially unexpected behavior of mixer), please attach the +output of this proc file together with the bug report. diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt new file mode 100644 index 00000000..fea65bb6 --- /dev/null +++ b/Documentation/sound/alsa/ControlNames.txt @@ -0,0 +1,85 @@ +This document describes standard names of mixer controls. + +Syntax: SOURCE [DIRECTION] FUNCTION + +DIRECTION: + (both directions) + Playback + Capture + Bypass Playback + Bypass Capture + +FUNCTION: + Switch (on/off switch) + Volume + Route (route control, hardware specific) + +SOURCE: + Master + Master Mono + Hardware Master + Speaker (internal speaker) + Headphone + Beep (beep generator) + Phone + Phone Input + Phone Output + Synth + FM + Mic + Line + CD + Video + Zoom Video + Aux + PCM + PCM Front + PCM Rear + PCM Pan + Loopback + Analog Loopback (D/A -> A/D loopback) + Digital Loopback (playback -> capture loopback - without analog path) + Mono + Mono Output + Multi + ADC + Wave + Music + I2S + IEC958 + +Exceptions: + [Digital] Capture Source + [Digital] Capture Switch (aka input gain switch) + [Digital] Capture Volume (aka input gain volume) + [Digital] Playback Switch (aka output gain switch) + [Digital] Playback Volume (aka output gain volume) + Tone Control - Switch + Tone Control - Bass + Tone Control - Treble + 3D Control - Switch + 3D Control - Center + 3D Control - Depth + 3D Control - Wide + 3D Control - Space + 3D Control - Level + Mic Boost [(?dB)] + +PCM interface: + + Sample Clock Source { "Word", "Internal", "AutoSync" } + Clock Sync Status { "Lock", "Sync", "No Lock" } + External Rate /* external capture rate */ + Capture Rate /* capture rate taken from external source */ + +IEC958 (S/PDIF) interface: + + IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */ + IEC958 [...] [Playback|Capture] Volume /* digital volume control */ + IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */ + IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */ + IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */ + IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */ + IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */ + IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */ + IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */ diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt new file mode 100644 index 00000000..d70c93bd --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -0,0 +1,428 @@ + 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 + w810 3-jack + z71v 3-jack (HP shared SPDIF) + asus 3-jack (ASUS Mobo) + asus-w1v ASUS W1V + asus-dig ASUS with SPDIF out + asus-dig2 ASUS with SPDIF out (using GPIO2) + uniwill 3-jack + fujitsu Fujitsu Laptops (Pi1536) + F1734 2-jack + lg LG laptop (m1 express dual) + lg-lw LG LW20/LW25 laptop + tcl TCL S700 + clevo Clevo laptops (m520G, m665n) + medion Medion Rim 2150 + test for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC260 +====== + hp HP machines + hp-3013 HP machines (3013-variant) + hp-dc7600 HP DC7600 + fujitsu Fujitsu S7020 + acer Acer TravelMate + will Will laptops (PB V7900) + replacer Replacer 672V + favorit100 Maxdata Favorit 100XS + basic fixed pin assignment (old default model) + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC262 +====== + fujitsu Fujitsu Laptop + hp-bpc HP xw4400/6400/8400/9400 laptops + hp-bpc-d7000 HP BPC D7000 + hp-tc-t5735 HP Thin Client T5735 + hp-rp5700 HP RP5700 + benq Benq ED8 + benq-t31 Benq T31 + hippo Hippo (ATI) with jack detection, Sony UX-90s + hippo_1 Hippo (Benq) with jack detection + sony-assamd Sony ASSAMD + toshiba-s06 Toshiba S06 + toshiba-rx1 Toshiba RX1 + tyan Tyan Thunder n6650W (S2915-E) + ultra Samsung Q1 Ultra Vista model + lenovo-3000 Lenovo 3000 y410 + nec NEC Versa S9100 + basic fixed pin assignment w/o SPDIF + auto auto-config reading BIOS (default) + +ALC267/268 +========== + quanta-il1 Quanta IL1 mini-notebook + 3stack 3-stack model + toshiba Toshiba A205 + acer Acer laptops + acer-dmic Acer laptops with digital-mic + acer-aspire Acer Aspire One + dell Dell OEM laptops (Vostro 1200) + zepto Zepto laptops + test for testing/debugging purpose, almost all controls can + adjusted. Appearing only when compiled with + $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) + +ALC269 +====== + basic Basic preset + quanta Quanta FL1 + laptop-amic Laptops with analog-mic input + laptop-dmic Laptops with digital-mic input + fujitsu FSC Amilo + lifebook Fujitsu Lifebook S6420 + auto auto-config reading BIOS (default) + +ALC662/663/272 +============== + 3stack-dig 3-stack (2-channel) with SPDIF + 3stack-6ch 3-stack (6-channel) + 3stack-6ch-dig 3-stack (6-channel) with SPDIF + 5stack-dig 5-stack with SPDIF + lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc P701 + eeepc-ep20 ASUS Eeepc EP20 + ecs ECS/Foxconn mobo + m51va ASUS M51VA + g71v ASUS G71V + h13 ASUS H13 + g50v ASUS G50V + asus-mode1 ASUS + asus-mode2 ASUS + asus-mode3 ASUS + asus-mode4 ASUS + asus-mode5 ASUS + asus-mode6 ASUS + asus-mode7 ASUS + asus-mode8 ASUS + dell Dell with ALC272 + dell-zm1 Dell ZM1 with ALC272 + samsung-nc10 Samsung NC10 mini notebook + auto auto-config reading BIOS (default) + +ALC680 +====== + base Base model (ASUS NX90) + auto auto-config reading BIOS (default) + +ALC882/883/885/888/889 +====================== + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack digital with SPDIF I/O + arima Arima W820Di1 + targa Targa T8, MSI-1049 T8 + asus-a7j ASUS A7J + asus-a7m ASUS A7M + macpro MacPro support + mb5 Macbook 5,1 + macmini3 Macmini 3,1 + mba21 Macbook Air 2,1 + mbp3 Macbook Pro rev3 + imac24 iMac 24'' with jack detection + imac91 iMac 9,1 + w2jc ASUS W2JC + 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) + alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) + 3stack-6ch 3-jack 6-channel + 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O + 6stack-dig-demo 6-jack digital for Intel demo board + acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) + acer-aspire Acer Aspire 9810 + acer-aspire-4930g Acer Aspire 4930G + acer-aspire-6530g Acer Aspire 6530G + acer-aspire-7730g Acer Aspire 7730G + acer-aspire-8930g Acer Aspire 8930G + medion Medion Laptops + targa-dig Targa/MSI + targa-2ch-dig Targa/MSI with 2-channel + targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) + laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) + lenovo-101e Lenovo 101E + lenovo-nb0763 Lenovo NB0763 + lenovo-ms7195-dig Lenovo MS7195 + lenovo-sky Lenovo Sky + haier-w66 Haier W66 + 3stack-hp HP machines with 3stack (Lucknow, Samba boards) + 6stack-dell Dell machines with 6stack (Inspiron 530) + mitac Mitac 8252D + clevo-m540r Clevo M540R (6ch + digital) + clevo-m720 Clevo M720 laptop series + fujitsu-pi2515 Fujitsu AMILO Pi2515 + fujitsu-xa3530 Fujitsu AMILO XA3530 + 3stack-6ch-intel Intel DG33* boards + intel-alc889a Intel IbexPeak with ALC889A + intel-x58 Intel DX58 with ALC889 + asus-p5q ASUS P5Q-EM boards + mb31 MacBook 3,1 + sony-vaio-tt Sony VAIO TT + auto auto-config reading BIOS (default) + +ALC861/660 +========== + 3stack 3-jack + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack with SPDIF I/O + 3stack-660 3-jack (for ALC660) + uniwill-m31 Uniwill M31 laptop + toshiba Toshiba laptop support + asus Asus laptop support + asus-laptop ASUS F2/F3 laptops + auto auto-config reading BIOS (default) + +ALC861VD/660VD +============== + 3stack 3-jack + 3stack-dig 3-jack with SPDIF OUT + 6stack-dig 6-jack with SPDIF OUT + 3stack-660 3-jack (for ALC660VD) + 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD) + lenovo Lenovo 3000 C200 + dallas Dallas laptops + hp HP TX1000 + asus-v1s ASUS V1Sn + auto auto-config reading BIOS (default) + +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 (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 +======= + 6stack 6-jack, separate surrounds (default) + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + laptop-eapd 2-channel with EAPD (ASUS A6J) + laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) + ultra 2-channel with EAPD (Samsung Ultra tablet PC) + samsung 2-channel with EAPD (Samsung R65) + samsung-p50 2-channel with HP-automute (Samsung P50) + +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 + lenovo-x200 Lenovo X200 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 + 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 + hp HP laptops with (inverted) mute-LED + hp-dv7-4000 HP dv-7 4000 + auto BIOS setup (default) + +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) + +VIA VT17xx/VT18xx/VT20xx +======================== + auto BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt new file mode 100644 index 00000000..c82beb00 --- /dev/null +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -0,0 +1,723 @@ +MORE NOTES ON HD-AUDIO DRIVER +============================= + Takashi Iwai + + +GENERAL +------- + +HD-audio is the new standard on-board audio component on modern PCs +after AC97. Although Linux has been supporting HD-audio since long +time ago, there are often problems with new machines. A part of the +problem is broken BIOS, and the rest is the driver implementation. +This document explains the brief trouble-shooting and debugging +methods for the HD-audio hardware. + +The HD-audio component consists of two parts: the controller chip and +the codec chips on the HD-audio bus. Linux provides a single driver +for all controllers, snd-hda-intel. Although the driver name contains +a word of a well-known hardware vendor, it's not specific to it but for +all controller chips by other companies. Since the HD-audio +controllers are supposed to be compatible, the single snd-hda-driver +should work in most cases. But, not surprisingly, there are known +bugs and issues specific to each controller type. The snd-hda-intel +driver has a bunch of workarounds for these as described below. + +A controller may have multiple codecs. Usually you have one audio +codec and optionally one modem codec. In theory, there might be +multiple audio codecs, e.g. for analog and digital outputs, and the +driver might not work properly because of conflict of mixer elements. +This should be fixed in future if such hardware really exists. + +The snd-hda-intel driver has several different codec parsers depending +on the codec. It has a generic parser as a fallback, but this +functionality is fairly limited until now. Instead of the generic +parser, usually the codec-specific parser (coded in patch_*.c) is used +for the codec-specific implementations. The details about the +codec-specific problems are explained in the later sections. + +If you are interested in the deep debugging of HD-audio, read the +HD-audio specification at first. The specification is found on +Intel's web page, for example: + +- http://www.intel.com/standards/hdaudio/ + + +HD-AUDIO CONTROLLER +------------------- + +DMA-Position Problem +~~~~~~~~~~~~~~~~~~~~ +The most common problem of the controller is the inaccurate DMA +pointer reporting. The DMA pointer for playback and capture can be +read in two ways, either via a LPIB register or via a position-buffer +map. As default the driver tries to read from the io-mapped +position-buffer, and falls back to LPIB if the position-buffer appears +dead. However, this detection isn't perfect on some devices. In such +a case, you can change the default method via `position_fix` option. + +`position_fix=1` means to use LPIB method explicitly. +`position_fix=2` means to use the position-buffer. +`position_fix=3` means to use a combination of both methods, needed +for some VIA and ATI controllers. 0 is the default value for all other +controllers, the automatic check and fallback to LPIB as described in +the above. If you get a problem of repeated sounds, this option might +help. + +In addition to that, every controller is known to be broken regarding +the wake-up timing. It wakes up a few samples before actually +processing the data on the buffer. This caused a lot of problems, for +example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts +an artificial delay to the wake up timing. This delay is controlled +via `bdl_pos_adj` option. + +When `bdl_pos_adj` is a negative value (as default), it's assigned to +an appropriate value depending on the controller chip. For Intel +chips, it'd be 1 while it'd be 32 for others. Usually this works. +Only in case it doesn't work and you get warning messages, you should +change this parameter to other values. + + +Codec-Probing Problem +~~~~~~~~~~~~~~~~~~~~~ +A less often but a more severe problem is the codec probing. When +BIOS reports the available codec slots wrongly, the driver gets +confused and tries to access the non-existing codec slot. This often +results in the total screw-up, and destructs the further communication +with the codec chips. The symptom appears usually as error messages +like: +------------------------------------------------------------------------ + hda_intel: azx_get_response timeout, switching to polling mode: + last cmd=0x12345678 + hda_intel: azx_get_response timeout, switching to single_cmd mode: + last cmd=0x12345678 +------------------------------------------------------------------------ + +The first line is a warning, and this is usually relatively harmless. +It means that the codec response isn't notified via an IRQ. The +driver uses explicit polling method to read the response. It gives +very slight CPU overhead, but you'd unlikely notice it. + +The second line is, however, a fatal error. If this happens, usually +it means that something is really wrong. Most likely you are +accessing a non-existing codec slot. + +Thus, if the second error message appears, try to narrow the probed +codec slots via `probe_mask` option. It's a bitmask, and each bit +corresponds to the codec slot. For example, to probe only the first +slot, pass `probe_mask=1`. For the first and the third slots, pass +`probe_mask=5` (where 5 = 1 | 4), and so on. + +Since 2.6.29 kernel, the driver has a more robust probing method, so +this error might happen rarely, though. + +On a machine with a broken BIOS, sometimes you need to force the +driver to probe the codec slots the hardware doesn't report for use. +In such a case, turn the bit 8 (0x100) of `probe_mask` option on. +Then the rest 8 bits are passed as the codec slots to probe +unconditionally. For example, `probe_mask=0x103` will force to probe +the codec slots 0 and 1 no matter what the hardware reports. + + +Interrupt Handling +~~~~~~~~~~~~~~~~~~ +HD-audio driver uses MSI as default (if available) since 2.6.33 +kernel as MSI works better on some machines, and in general, it's +better for performance. However, Nvidia controllers showed bad +regressions with MSI (especially in a combination with AMD chipset), +thus we disabled MSI for them. + +There seem also still other devices that don't work with MSI. If you +see a regression wrt the sound quality (stuttering, etc) or a lock-up +in the recent kernel, try to pass `enable_msi=0` option to disable +MSI. If it works, you can add the known bad device to the blacklist +defined in hda_intel.c. In such a case, please report and give the +patch back to the upstream developer. + + +HD-AUDIO CODEC +-------------- + +Model Option +~~~~~~~~~~~~ +The most common problem regarding the HD-audio driver is the +unsupported codec features or the mismatched device configuration. +Most of codec-specific code has several preset models, either to +override the BIOS setup or to provide more comprehensive features. + +The driver checks PCI SSID and looks through the static configuration +table until any matching entry is found. If you have a new machine, +you may see a message like below: +------------------------------------------------------------------------ + hda_codec: ALC880: BIOS auto-probing. +------------------------------------------------------------------------ +Meanwhile, in the earlier versions, you would see a message like: +------------------------------------------------------------------------ + hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... +------------------------------------------------------------------------ +Even if you see such a message, DON'T PANIC. Take a deep breath and +keep your towel. First of all, it's an informational message, no +warning, no error. This means that the PCI SSID of your device isn't +listed in the known preset model (white-)list. But, this doesn't mean +that the driver is broken. Many codec-drivers provide the automatic +configuration mechanism based on the BIOS setup. + +The HD-audio codec has usually "pin" widgets, and BIOS sets the default +configuration of each pin, which indicates the location, the +connection type, the jack color, etc. The HD-audio driver can guess +the right connection judging from these default configuration values. +However -- some codec-support codes, such as patch_analog.c, don't +support the automatic probing (yet as of 2.6.28). And, BIOS is often, +yes, pretty often broken. It sets up wrong values and screws up the +driver. + +The preset model is provided basically to overcome such a situation. +When the matching preset model is found in the white-list, the driver +assumes the static configuration of that preset and builds the mixer +elements and PCM streams based on the static information. Thus, if +you have a newer machine with a slightly different PCI SSID from the +existing one, you may have a good chance to re-use the same model. +You can pass the `model` option to specify the preset model instead of +PCI SSID look-up. + +What `model` option values are available depends on the codec chip. +Check your codec chip from the codec proc file (see "Codec Proc-File" +section below). It will show the vendor/product name of your codec +chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file, +the section of HD-audio driver. You can find a list of codecs +and `model` options belonging to each codec. For example, for Realtek +ALC262 codec chip, pass `model=ultra` for devices that are compatible +with Samsung Q1 Ultra. + +Thus, the first thing you can do for any brand-new, unsupported and +non-working HD-audio hardware is to check HD-audio codec and several +different `model` option values. If you have any luck, some of them +might suit with your device well. + +Some codecs such as ALC880 have a special model option `model=test`. +This configures the driver to provide as many mixer controls as +possible for every single pin feature except for the unsolicited +events (and maybe some other specials). Adjust each mixer element and +try the I/O in the way of trial-and-error until figuring out the whole +I/O pin mappings. + +Note that `model=generic` has a special meaning. It means to use the +generic parser regardless of the codec. Usually the codec-specific +parser is much better than the generic parser (as now). Thus this +option is more about the debugging purpose. + +Speaker and Headphone Output +~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +One of the most frequent (and obvious) bugs with HD-audio is the +silent output from either or both of a built-in speaker and a +headphone jack. In general, you should try a headphone output at +first. A speaker output often requires more additional controls like +the external amplifier bits. Thus a headphone output has a slightly +better chance. + +Before making a bug report, double-check whether the mixer is set up +correctly. The recent version of snd-hda-intel driver provides mostly +"Master" volume control as well as "Front" volume (where Front +indicates the front-channels). In addition, there can be individual +"Headphone" and "Speaker" controls. + +Ditto for the speaker output. There can be "External Amplifier" +switch on some codecs. Turn on this if present. + +Another related problem is the automatic mute of speaker output by +headphone plugging. This feature is implemented in most cases, but +not on every preset model or codec-support code. + +In anyway, try a different model option if you have such a problem. +Some other models may match better and give you more matching +functionality. If none of the available models works, send a bug +report. See the bug report section for details. + +If you are masochistic enough to debug the driver problem, note the +following: + +- The speaker (and the headphone, too) output often requires the + external amplifier. This can be set usually via EAPD verb or a + certain GPIO. If the codec pin supports EAPD, you have a better + chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly + it's either GPIO0 or GPIO1) may turn on/off EAPD. +- Some Realtek codecs require special vendor-specific coefficients to + turn on the amplifier. See patch_realtek.c. +- IDT codecs may have extra power-enable/disable controls on each + analog pin. See patch_sigmatel.c. +- Very rare but some devices don't accept the pin-detection verb until + triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the + codec-communication stall. Some examples are found in + patch_realtek.c. + + +Capture Problems +~~~~~~~~~~~~~~~~ +The capture problems are often because of missing setups of mixers. +Thus, before submitting a bug report, make sure that you set up the +mixer correctly. For example, both "Capture Volume" and "Capture +Switch" have to be set properly in addition to the right "Capture +Source" or "Input Source" selection. Some devices have "Mic Boost" +volume or switch. + +When the PCM device is opened via "default" PCM (without pulse-audio +plugin), you'll likely have "Digital Capture Volume" control as well. +This is provided for the extra gain/attenuation of the signal in +software, especially for the inputs without the hardware volume +control such as digital microphones. Unless really needed, this +should be set to exactly 50%, corresponding to 0dB -- neither extra +gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM, +this control will have no influence, though. + +It's known that some codecs / devices have fairly bad analog circuits, +and the recorded sound contains a certain DC-offset. This is no bug +of the driver. + +Most of modern laptops have no analog CD-input connection. Thus, the +recording from CD input won't work in many cases although the driver +provides it as the capture source. Use CDDA instead. + +The automatic switching of the built-in and external mic per plugging +is implemented on some codec models but not on every model. Partly +because of my laziness but mostly lack of testers. Feel free to +submit the improvement patch to the author. + + +Direct Debugging +~~~~~~~~~~~~~~~~ +If no model option gives you a better result, and you are a tough guy +to fight against evil, try debugging via hitting the raw HD-audio +codec verbs to the device. Some tools are available: hda-emu and +hda-analyzer. The detailed description is found in the sections +below. You'd need to enable hwdep for using these tools. See "Kernel +Configuration" section. + + +OTHER ISSUES +------------ + +Kernel Configuration +~~~~~~~~~~~~~~~~~~~~ +In general, I recommend you to enable the sound debug option, +`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not. +This enables snd_printd() macro and others, and you'll get additional +kernel messages at probing. + +In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this +will give you far more messages. Thus turn this on only when you are +sure to want it. + +Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*` +options. Note that each of them corresponds to the codec chip, not +the controller chip. Thus, even if lspci shows the Nvidia controller, +you may need to choose the option for other vendors. If you are +unsure, just select all yes. + +`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver. +When this is enabled, the driver creates hardware-dependent devices +(one per each codec), and you have a raw access to the device via +these device files. For example, `hwC0D2` will be created for the +codec slot #2 of the first card (#0). For debug-tools such as +hda-verb and hda-analyzer, the hwdep device has to be enabled. +Thus, it'd be better to turn this on always. + +`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the +hwdep option above. When enabled, you'll have some sysfs files under +the corresponding hwdep directory. See "HD-audio reconfiguration" +section below. + +`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature. +See "Power-saving" section below. + + +Codec Proc-File +~~~~~~~~~~~~~~~ +The codec proc-file is a treasure-chest for debugging HD-audio. +It shows most of useful information of each codec widget. + +The proc file is located in /proc/asound/card*/codec#*, one file per +each codec slot. You can know the codec vendor, product id and +names, the type of each widget, capabilities and so on. +This file, however, doesn't show the jack sensing state, so far. This +is because the jack-sensing might be depending on the trigger state. + +This file will be picked up by the debug tools, and also it can be fed +to the emulator as the primary codec information. See the debug tools +section below. + +This proc file can be also used to check whether the generic parser is +used. When the generic parser is used, the vendor/product ID name +will appear as "Realtek ID 0262", instead of "Realtek ALC262". + + +HD-Audio Reconfiguration +~~~~~~~~~~~~~~~~~~~~~~~~ +This is an experimental feature to allow you re-configure the HD-audio +codec dynamically without reloading the driver. The following sysfs +files are available under each codec-hwdep device directory (e.g. +/sys/class/sound/hwC0D0): + +vendor_id:: + Shows the 32bit codec vendor-id hex number. You can change the + vendor-id value by writing to this file. +subsystem_id:: + Shows the 32bit codec subsystem-id hex number. You can change the + subsystem-id value by writing to this file. +revision_id:: + Shows the 32bit codec revision-id hex number. You can change the + revision-id value by writing to this file. +afg:: + Shows the AFG ID. This is read-only. +mfg:: + Shows the MFG ID. This is read-only. +name:: + Shows the codec name string. Can be changed by writing to this + file. +modelname:: + Shows the currently set `model` option. Can be changed by writing + to this file. +init_verbs:: + The extra verbs to execute at initialization. You can add a verb by + writing to this file. Pass three numbers: nid, verb and parameter + (separated with a space). +hints:: + Shows / stores hint strings for codec parsers for any use. + Its format is `key = value`. For example, passing `hp_detect = yes` + to IDT/STAC codec parser will result in the disablement of the + headphone detection. +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 +------------------------------------------------------------------------ + + +Early Patching +~~~~~~~~~~~~~~ +When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a +firmware file for modifying the HD-audio setup before initializing the +codec. This can work basically like the reconfiguration via sysfs in +the above, but it does it before the first codec configuration. + +A patch file is a plain text file which looks like below: + +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [model] + auto + + [pincfg] + 0x12 0x411111f0 + + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff + + [hint] + hp_detect = yes +------------------------------------------------------------------------ + +The file needs to have a line `[codec]`. The next line should contain +three numbers indicating the codec vendor-id (0x12345678 in the +example), the codec subsystem-id (0xabcd1234) and the address (2) of +the codec. The rest patch entries are applied to this specified codec +until another codec entry is given. + +The `[model]` line allows to change the model name of the each codec. +In the example above, it will be changed to model=auto. +Note that this overrides the module option. + +After the `[pincfg]` line, the contents are parsed as the initial +default pin-configurations just like `user_pin_configs` sysfs above. +The values can be shown in user_pin_configs sysfs file, too. + +Similarly, the lines after `[verb]` are parsed as `init_verbs` +sysfs entries, and the lines after `[hint]` are parsed as `hints` +sysfs entries, respectively. + +Another example to override the codec vendor id from 0x12345678 to +0xdeadbeef is like below: +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [vendor_id] + 0xdeadbeef +------------------------------------------------------------------------ + +In the similar way, you can override the codec subsystem_id via +`[subsystem_id]`, the revision id via `[revision_id]` line. +Also, the codec chip name can be rewritten via `[chip_name]` line. +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [subsystem_id] + 0xffff1111 + + [revision_id] + 0x10 + + [chip_name] + My-own NEWS-0002 +------------------------------------------------------------------------ + +The hd-audio driver reads the file via request_firmware(). Thus, +a patch file has to be located on the appropriate firmware path, +typically, /lib/firmware. For example, when you pass the option +`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be +present. + +The patch module option is specific to each card instance, and you +need to give one file name for each instance, separated by commas. +For example, if you have two cards, one for an on-board analog and one +for an HDMI video board, you may pass patch option like below: +------------------------------------------------------------------------ + options snd-hda-intel patch=on-board-patch,hdmi-patch +------------------------------------------------------------------------ + + +Power-Saving +~~~~~~~~~~~~ +The power-saving is a kind of auto-suspend of the device. When the +device is inactive for a certain time, the device is automatically +turned off to save the power. The time to go down is specified via +`power_save` module option, and this option can be changed dynamically +via sysfs. + +The power-saving won't work when the analog loopback is enabled on +some codecs. Make sure that you mute all unneeded signal routes when +you want the power-saving. + +The power-saving feature might cause audible click noises at each +power-down/up depending on the device. Some of them might be +solvable, but some are hard, I'm afraid. Some distros such as +openSUSE enables the power-saving feature automatically when the power +cable is unplugged. Thus, if you hear noises, suspect first the +power-saving. See /sys/module/snd_hda_intel/parameters/power_save to +check the current value. If it's non-zero, the feature is turned on. + + +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-2.6.git + +The master branch or for-next branches can be used as the main +development branches in general while the HD-audio specific patches +are committed in topic/hda branch. + +If you are using the latest Linus tree, it'd be better to pull the +above GIT tree onto it. If you are using the older kernels, an easy +way to try the latest ALSA code is to build from the snapshot +tarball. There are daily tarballs and the latest snapshot tarball. +All can be built just like normal alsa-driver release packages, that +is, installed via the usual spells: configure, make and make +install(-modules). See INSTALL in the package. The snapshot tarballs +are found at: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ + + +Sending a Bug Report +~~~~~~~~~~~~~~~~~~~~ +If any model or module options don't work for your device, it's time +to send a bug report to the developers. Give the following in your +bug report: + +- Hardware vendor, product and model names +- Kernel version (and ALSA-driver version if you built externally) +- `alsa-info.sh` output; run with `--no-upload` option. See the + section below about alsa-info + +If it's a regression, at best, send alsa-info outputs of both working +and non-working kernels. This is really helpful because we can +compare the codec registers directly. + +Send a bug report either the followings: + +kernel-bugzilla:: + https://bugzilla.kernel.org/ +alsa-devel ML:: + alsa-devel@alsa-project.org + + +DEBUG TOOLS +----------- + +This section describes some tools available for debugging HD-audio +problems. + +alsa-info +~~~~~~~~~ +The script `alsa-info.sh` is a very useful tool to gather the audio +device information. You can fetch the latest version from: + +- http://www.alsa-project.org/alsa-info.sh + +Run this script as root, and it will gather the important information +such as the module lists, module parameters, proc file contents +including the codec proc files, mixer outputs and the control +elements. As default, it will store the information onto a web server +on alsa-project.org. But, if you send a bug report, it'd be better to +run with `--no-upload` option, and attach the generated file. + +There are some other useful options. See `--help` option output for +details. + +When a probe error occurs or when the driver obviously assigns a +mismatched model, it'd be helpful to load the driver with +`probe_only=1` option (at best after the cold reboot) and run +alsa-info at this state. With this option, the driver won't configure +the mixer and PCM but just tries to probe the codec slot. After +probing, the proc file is available, so you can get the raw codec +information before modified by the driver. Of course, the driver +isn't usable with `probe_only=1`. But you can continue the +configuration via hwdep sysfs file if hda-reconfig option is enabled. +Using `probe_only` mask 2 skips the reset of HDA codecs (use +`probe_only=3` as module option). The hwdep interface can be used +to determine the BIOS codec initialization. + + +hda-verb +~~~~~~~~ +hda-verb is a tiny program that allows you to access the HD-audio +codec directly. You can execute a raw HD-audio codec verb with this. +This program accesses the hwdep device, thus you need to enable the +kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand. + +The hda-verb program takes four arguments: the hwdep device file, the +widget NID, the verb and the parameter. When you access to the codec +on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first +argument, typically. (However, the real path name depends on the +system.) + +The second parameter is the widget number-id to access. The third +parameter can be either a hex/digit number or a string corresponding +to a verb. Similarly, the last parameter is the value to write, or +can be a string for the parameter type. + +------------------------------------------------------------------------ + % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 + nid = 0x12, verb = 0x701, param = 0x2 + value = 0x0 + + % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID + nid = 0x0, verb = 0xf00, param = 0x0 + value = 0x10ec0262 + + % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 + nid = 0x2, verb = 0x300, param = 0xb080 + value = 0x0 +------------------------------------------------------------------------ + +Although you can issue any verbs with this program, the driver state +won't be always updated. For example, the volume values are usually +cached in the driver, and thus changing the widget amp value directly +via hda-verb won't change the mixer value. + +The hda-verb program is found in the ftp directory: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ + +Also a git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git + +See README file in the tarball for more details about hda-verb +program. + + +hda-analyzer +~~~~~~~~~~~~ +hda-analyzer provides a graphical interface to access the raw HD-audio +control, based on pyGTK2 binding. It's a more powerful version of +hda-verb. The program gives you an easy-to-use GUI stuff for showing +the widget information and adjusting the amp values, as well as the +proc-compatible output. + +The hda-analyzer: + +- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer + +is a part of alsa.git repository in alsa-project.org: + +- git://git.alsa-project.org/alsa.git + +Codecgraph +~~~~~~~~~~ +Codecgraph is a utility program to generate a graph and visualizes the +codec-node connection of a codec chip. It's especially useful when +you analyze or debug a codec without a proper datasheet. The program +parses the given codec proc file and converts to SVG via graphiz +program. + +The tarball and GIT trees are found in the web page at: + +- http://helllabs.org/codecgraph/ + + +hda-emu +~~~~~~~ +hda-emu is an HD-audio emulator. The main purpose of this program is +to debug an HD-audio codec without the real hardware. Thus, it +doesn't emulate the behavior with the real audio I/O, but it just +dumps the codec register changes and the ALSA-driver internal changes +at probing and operating the HD-audio driver. + +The program requires a codec proc-file to simulate. Get a proc file +for the target codec beforehand, or pick up an example codec from the +codec proc collections in the tarball. Then, run the program with the +proc file, and the hda-emu program will start parsing the codec file +and simulates the HD-audio driver: + +------------------------------------------------------------------------ + % hda-emu codecs/stac9200-dell-d820-laptop + # Parsing.. + hda_codec: Unknown model for STAC9200, using BIOS defaults + hda_codec: pin nid 08 bios pin config 40c003fa + .... +------------------------------------------------------------------------ + +The program gives you only a very dumb command-line interface. You +can get a proc-file dump at the current state, get a list of control +(mixer) elements, set/get the control element value, simulate the PCM +operation, the jack plugging simulation, etc. + +The package is found in: + +- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ + +A git repository is available: + +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git + +See README file in the tarball for more details about hda-emu +program. diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt new file mode 100644 index 00000000..ccda41b1 --- /dev/null +++ b/Documentation/sound/alsa/Joystick.txt @@ -0,0 +1,86 @@ +Analog Joystick Support on ALSA Drivers +======================================= + Oct. 14, 2003 + Takashi Iwai + +General +------- + +First of all, you need to enable GAMEPORT support on Linux kernel for +using a joystick with the ALSA driver. For the details of gameport +support, refer to Documentation/input/joystick.txt. + +The joystick support of ALSA drivers is different between ISA and PCI +cards. In the case of ISA (PnP) cards, it's usually handled by the +independent module (ns558). Meanwhile, the ALSA PCI drivers have the +built-in gameport support. Hence, when the ALSA PCI driver is built +in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the +gameport support on that card will be (silently) disabled. + +Some adapter modules probe the physical connection of the device at +the load time. It'd be safer to plug in the joystick device before +loading the module. + + +PCI Cards +--------- + +For PCI cards, the joystick is enabled when the appropriate module +option is specified. Some drivers don't need options, and the +joystick support is always enabled. In the former ALSA version, there +was a dynamic control API for the joystick activation. It was +changed, however, to the static module options because of the system +stability and the resource management. + +The following PCI drivers support the joystick natively. + + Driver Module Option Available Values + --------------------------------------------------------------------------- + als4000 joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + au88x0 N/A N/A + azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) + ens1370 joystick 0 = disable (default), 1 = enable + ens1371 joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x200, 0x208, 0x210, 0x218 + cmipci joystick_port 0 = disable (default), 1 = auto-detect, + manual: any address (e.g. 0x200) + cs4281 N/A N/A + cs46xx N/A N/A + es1938 N/A N/A + es1968 joystick 0 = disable (default), 1 = enable + sonicvibes N/A N/A + trident N/A N/A + via82xx(*1) joystick 0 = disable (default), 1 = enable + ymfpci joystick_port 0 = disable (default), 1 = auto-detect, + manual: 0x201, 0x202, 0x204, 0x205(*2) + --------------------------------------------------------------------------- + + *1) VIA686A/B only + *2) With YMF744/754 chips, the port address can be chosen arbitrarily + +The following drivers don't support gameport natively, but there are +additional modules. Load the corresponding module to add the gameport +support. + + Driver Additional Module + ----------------------------- + emu10k1 emu10k1-gp + fm801 fm801-gp + ----------------------------- + +Note: the "pcigame" and "cs461x" modules are for the OSS drivers only. + These ALSA drivers (cs46xx, trident and au88x0) have the + built-in gameport support. + +As mentioned above, ALSA PCI drivers have the built-in gameport +support, so you don't have to load ns558 module. Just load "joydev" +and the appropriate adapter module (e.g. "analog"). + + +ISA Cards +--------- + +ALSA ISA drivers don't have the built-in gameport support. +Instead, you need to load "ns558" module in addition to "joydev" and +the adapter module (e.g. "analog"). diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt new file mode 100644 index 00000000..ef42c44f --- /dev/null +++ b/Documentation/sound/alsa/MIXART.txt @@ -0,0 +1,100 @@ + Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards + Digigram + + +GENERAL +======= + +The miXart8 is a multichannel audio processing and mixing soundcard +that has 4 stereo audio inputs and 4 stereo audio outputs. +The miXart8AES/EBU is the same with a add-on card that offers further +4 digital stereo audio inputs and outputs. +Furthermore the add-on card offers external clock synchronisation +(AES/EBU, Word Clock, Time Code and Video Synchro) + +The mainboard has a PowerPC that offers onboard mpeg encoding and +decoding, samplerate conversions and various effects. + +The driver don't work properly at all until the certain firmwares +are loaded, i.e. no PCM nor mixer devices will appear. +Use the mixartloader that can be found in the alsa-tools package. + + +VERSION 0.1.0 +============= + +One miXart8 board will be represented as 4 alsa cards, each with 1 +stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device. +With a miXart8AES/EBU there is in addition 1 stereo digital input +'pcm1c' and 1 stereo digital output 'pcm1p' per card. + +Formats +------- +U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE +Sample rates : 8000 - 48000 Hz continuously + +Playback +-------- +For instance the playback devices are configured to have max. 4 +substreams performing hardware mixing. This could be changed to a +maximum of 24 substreams if wished. +Mono files will be played on the left and right channel. Each channel +can be muted for each stream to use 8 analog/digital outputs separately. + +Capture +------- +There is one substream per capture device. For instance only stereo +formats are supported. + +Mixer +----- + and : analog volume control of playback and capture PCM. + and : digital volume control of each analog substream. + and : digital volume control of each AES/EBU substream. + : Loopback from 'pcm0c' to 'pcm0p' with digital volume +and mute control. + +Rem : for best audio quality try to keep a 0 attenuation on the PCM +and AES volume controls which is set by 219 in the range from 0 to 255 +(about 86% with alsamixer) + + +NOT YET IMPLEMENTED +=================== + +- external clock support (AES/EBU, Word Clock, Time Code, Video Sync) +- MPEG audio formats +- mono record +- on-board effects and samplerate conversions +- linked streams + + +FIRMWARE +======== + +[As of 2.6.11, the firmware can be loaded automatically with hotplug + when CONFIG_FW_LOADER is set. The mixartloader is necessary only + for older versions or when you build the driver into kernel.] + +For loading the firmware automatically after the module is loaded, use +the post-install command. For example, add the following entry to +/etc/modprobe.conf for miXart driver: + + install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ + /usr/bin/mixartloader +(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to + /etc/modules.conf, instead.) + +The firmware binaries are installed on /usr/share/alsa/firmware +(or /usr/local/share/alsa/firmware, depending to the prefix option of +configure). There will be a miXart.conf file, which define the dsp image +files. + +The firmware files are copyright by Digigram SA + + +COPYRIGHT +========= + +Copyright (c) 2003 Digigram SA +Distributable under GPL. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt new file mode 100644 index 00000000..022aaeb0 --- /dev/null +++ b/Documentation/sound/alsa/OSS-Emulation.txt @@ -0,0 +1,305 @@ + NOTES ON KERNEL OSS-EMULATION + ============================= + + Jan. 22, 2004 Takashi Iwai + + +Modules +======= + +ALSA provides a powerful OSS emulation on the kernel. +The OSS emulation for PCM, mixer and sequencer devices is implemented +as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. +When you need to access the OSS PCM, mixer or sequencer devices, the +corresponding module has to be loaded. + +These modules are loaded automatically when the corresponding service +is called. The alias is defined sound-service-x-y, where x and y are +the card number and the minor unit number. Usually you don't have to +define these aliases by yourself. + +Only necessary step for auto-loading of OSS modules is to define the +card alias in /etc/modprobe.conf, such as + + alias sound-slot-0 snd-emu10k1 + +As the second card, define sound-slot-1 as well. +Note that you can't use the aliased name as the target name (i.e. +"alias sound-slot-0 snd-card-0" doesn't work any more like the old +modutils). + +The currently available OSS configuration is shown in +/proc/asound/oss/sndstat. This shows in the same syntax of +/dev/sndstat, which is available on the commercial OSS driver. +On ALSA, you can symlink /dev/sndstat to this proc file. + +Please note that the devices listed in this proc file appear only +after the corresponding OSS-emulation module is loaded. Don't worry +even if "NOT ENABLED IN CONFIG" is shown in it. + + +Device Mapping +============== + +ALSA supports the following OSS device files: + + PCM: + /dev/dspX + /dev/adspX + + Mixer: + /dev/mixerX + + MIDI: + /dev/midi0X + /dev/amidi0X + + Sequencer: + /dev/sequencer + /dev/sequencer2 (aka /dev/music) + +where X is the card number from 0 to 7. + +(NOTE: Some distributions have the device files like /dev/midi0 and + /dev/midi1. They are NOT for OSS but for tclmidi, which is + a totally different thing.) + +Unlike the real OSS, ALSA cannot use the device files more than the +assigned ones. For example, the first card cannot use /dev/dsp1 or +/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. + +As seen above, PCM and MIDI may have two devices. Usually, the first +PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary +device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and +/dev/amidi, respectively. + +You can change this device mapping via the module options of +snd-pcm-oss and snd-rawmidi. In the case of PCM, the following +options are available for snd-pcm-oss: + + dsp_map PCM device number assigned to /dev/dspX + (default = 0) + adsp_map PCM device number assigned to /dev/adspX + (default = 1) + +For example, to map the third PCM device (hw:0,2) to /dev/adsp0, +define like this: + + options snd-pcm-oss adsp_map=2 + +The options take arrays. For configuring the second card, specify +two entries separated by comma. For example, to map the third PCM +device on the second card to /dev/adsp1, define like below: + + options snd-pcm-oss adsp_map=0,2 + +To change the mapping of MIDI devices, the following options are +available for snd-rawmidi: + + midi_map MIDI device number assigned to /dev/midi0X + (default = 0) + amidi_map MIDI device number assigned to /dev/amidi0X + (default = 1) + +For example, to assign the third MIDI device on the first card to +/dev/midi00, define as follows: + + options snd-rawmidi midi_map=2 + + +PCM Mode +======== + +As default, ALSA emulates the OSS PCM with so-called plugin layer, +i.e. tries to convert the sample format, rate or channels +automatically when the card doesn't support it natively. +This will lead to some problems for some applications like quake or +wine, especially if they use the card only in the MMAP mode. + +In such a case, you can change the behavior of PCM per application by +writing a command to the proc file. There is a proc file for each PCM +stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number +(zero-based), Y the PCM device number (zero-based), and 'p' is for +playback and 'c' for capture, respectively. Note that this proc file +exists only after snd-pcm-oss module is loaded. + +The command sequence has the following syntax: + + app_name fragments fragment_size [options] + +app_name is the name of application with (higher priority) or without +path. +fragments specifies the number of fragments or zero if no specific +number is given. +fragment_size is the size of fragment in bytes or zero if not given. +options is the optional parameters. The following options are +available: + + disable the application tries to open a pcm device for + this channel but does not want to use it. + direct don't use plugins + block force block open mode + non-block force non-block open mode + partial-frag write also partial fragments (affects playback only) + no-silence do not fill silence ahead to avoid clicks + +The disable option is useful when one stream direction (playback or +capture) is not handled correctly by the application although the +hardware itself does support both directions. +The direct option is used, as mentioned above, to bypass the automatic +conversion and useful for MMAP-applications. +For example, to playback the first PCM device without plugins for +quake, send a command via echo like the following: + + % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss + +While quake wants only playback, you may append the second command +to notify driver that only this direction is about to be allocated: + + % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss + +The permission of proc files depend on the module options of snd. +As default it's set as root, so you'll likely need to be superuser for +sending the command above. + +The block and non-block options are used to change the behavior of +opening the device file. + +As default, ALSA behaves as original OSS drivers, i.e. does not block +the file when it's busy. The -EBUSY error is returned in this case. + +This blocking behavior can be changed globally via nonblock_open +module option of snd-pcm-oss. For using the blocking mode as default +for OSS devices, define like the following: + + options snd-pcm-oss nonblock_open=0 + +The partial-frag and no-silence commands have been added recently. +Both commands are for optimization use only. The former command +specifies to invoke the write transfer only when the whole fragment is +filled. The latter stops writing the silence data ahead +automatically. Both are disabled as default. + +You can check the currently defined configuration by reading the proc +file. The read image can be sent to the proc file again, hence you +can save the current configuration + + % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg + +and restore it like + + % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss + +Also, for clearing all the current configuration, send "erase" command +as below: + + % echo "erase" > /proc/asound/card0/pcm0p/oss + + +Mixer Elements +============== + +Since ALSA has completely different mixer interface, the emulation of +OSS mixer is relatively complicated. ALSA builds up a mixer element +from several different ALSA (mixer) controls based on the name +string. For example, the volume element SOUND_MIXER_PCM is composed +from "PCM Playback Volume" and "PCM Playback Switch" controls for the +playback direction and from "PCM Capture Volume" and "PCM Capture +Switch" for the capture directory (if exists). When the PCM volume of +OSS is changed, all the volume and switch controls above are adjusted +automatically. + +As default, ALSA uses the following control for OSS volumes: + + OSS volume ALSA control Index + ----------------------------------------------------- + SOUND_MIXER_VOLUME Master 0 + SOUND_MIXER_BASS Tone Control - Bass 0 + SOUND_MIXER_TREBLE Tone Control - Treble 0 + SOUND_MIXER_SYNTH Synth 0 + SOUND_MIXER_PCM PCM 0 + SOUND_MIXER_SPEAKER PC Speaker 0 + SOUND_MIXER_LINE Line 0 + SOUND_MIXER_MIC Mic 0 + SOUND_MIXER_CD CD 0 + SOUND_MIXER_IMIX Monitor Mix 0 + SOUND_MIXER_ALTPCM PCM 1 + SOUND_MIXER_RECLEV (not assigned) + SOUND_MIXER_IGAIN Capture 0 + SOUND_MIXER_OGAIN Playback 0 + SOUND_MIXER_LINE1 Aux 0 + SOUND_MIXER_LINE2 Aux 1 + SOUND_MIXER_LINE3 Aux 2 + SOUND_MIXER_DIGITAL1 Digital 0 + SOUND_MIXER_DIGITAL2 Digital 1 + SOUND_MIXER_DIGITAL3 Digital 2 + SOUND_MIXER_PHONEIN Phone 0 + SOUND_MIXER_PHONEOUT Phone 1 + SOUND_MIXER_VIDEO Video 0 + SOUND_MIXER_RADIO Radio 0 + SOUND_MIXER_MONITOR Monitor 0 + +The second column is the base-string of the corresponding ALSA +control. In fact, the controls with "XXX [Playback|Capture] +[Volume|Switch]" will be checked in addition. + +The current assignment of these mixer elements is listed in the proc +file, /proc/asound/cardX/oss_mixer, which will be like the following + + VOLUME "Master" 0 + BASS "" 0 + TREBLE "" 0 + SYNTH "" 0 + PCM "PCM" 0 + ... + +where the first column is the OSS volume element, the second column +the base-string of the corresponding ALSA control, and the third the +control index. When the string is empty, it means that the +corresponding OSS control is not available. + +For changing the assignment, you can write the configuration to this +proc file. For example, to map "Wave Playback" to the PCM volume, +send the command like the following: + + % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer + +The command is exactly as same as listed in the proc file. You can +change one or more elements, one volume per line. In the last +example, both "Wave Playback Volume" and "Wave Playback Switch" will +be affected when PCM volume is changed. + +Like the case of PCM proc file, the permission of proc files depend on +the module options of snd. you'll likely need to be superuser for +sending the command above. + +As well as in the case of PCM proc file, you can save and restore the +current mixer configuration by reading and writing the whole file +image. + + +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus + io_handle = open("device", O_RDWR) +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) +and set the values for the corresponding handle. + + +Unsupported Features +==================== + +MMAP on ICE1712 driver +---------------------- +ICE1712 supports only the unconventional format, interleaved +10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap +the buffer as the conventional (mono or 2-channels, 8 or 16bit) format +on OSS. + diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt new file mode 100644 index 00000000..7fcd1ad9 --- /dev/null +++ b/Documentation/sound/alsa/Procfile.txt @@ -0,0 +1,243 @@ + Proc Files of ALSA Drivers + ========================== + Takashi Iwai + +General +------- + +ALSA has its own proc tree, /proc/asound. Many useful information are +found in this tree. When you encounter a problem and need debugging, +check the files listed in the following sections. + +Each card has its subtree cardX, where X is from 0 to 7. The +card-specific files are stored in the card* subdirectories. + + +Global Information +------------------ + +cards + Shows the list of currently configured ALSA drivers, + index, the id string, short and long descriptions. + +version + Shows the version string and compile date. + +modules + Lists the module of each card + +devices + Lists the ALSA native device mappings. + +meminfo + Shows the status of allocated pages via ALSA drivers. + Appears only when CONFIG_SND_DEBUG=y. + +hwdep + Lists the currently available hwdep devices in format of + -: + +pcm + Lists the currently available PCM devices in format of + -: : : + +timer + Lists the currently available timer devices + + +oss/devices + Lists the OSS device mappings. + +oss/sndstat + Provides the output compatible with /dev/sndstat. + You can symlink this to /dev/sndstat. + + +Card Specific Files +------------------- + +The card-specific files are found in /proc/asound/card* directories. +Some drivers (e.g. cmipci) have their own proc entries for the +register dump, etc (e.g. /proc/asound/card*/cmipci shows the register +dump). These files would be really helpful for debugging. + +When PCM devices are available on this card, you can see directories +like pcm0p or pcm1c. They hold the PCM information for each PCM +stream. The number after 'pcm' is the PCM device number from 0, and +the last 'p' or 'c' means playback or capture direction. The files in +this subtree is described later. + +The status of MIDI I/O is found in midi* files. It shows the device +name and the received/transmitted bytes through the MIDI device. + +When the card is equipped with AC97 codecs, there are codec97#* +subdirectories (described later). + +When the OSS mixer emulation is enabled (and the module is loaded), +oss_mixer file appears here, too. This shows the current mapping of +OSS mixer elements to the ALSA control elements. You can change the +mapping by writing to this device. Read OSS-Emulation.txt for +details. + + +PCM Proc Files +-------------- + +card*/pcm*/info + The general information of this PCM device: card #, device #, + substreams, etc. + +card*/pcm*/xrun_debug + This file appears when CONFIG_SND_DEBUG=y and + CONFIG_PCM_XRUN_DEBUG=y. + This shows the status of xrun (= buffer overrun/xrun) and + invalid PCM position debug/check of ALSA PCM middle layer. + It takes an integer value, can be changed by writing to this + file, such as + + # echo 5 > /proc/asound/card0/pcm0p/xrun_debug + + The value consists of the following bit flags: + bit 0 = Enable XRUN/jiffies debug messages + bit 1 = Show stack trace at XRUN / jiffies check + bit 2 = Enable additional jiffies check + bit 3 = Log hwptr update at each period interrupt + bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr() + bit 5 = Show last 10 positions on error + bit 6 = Do above only once + + When the bit 0 is set, the driver will show the messages to + kernel log when an xrun is detected. The debug message is + shown also when the invalid H/W pointer is detected at the + update of periods (usually called from the interrupt + handler). + + When the bit 1 is set, the driver will show the stack trace + additionally. This may help the debugging. + + Since 2.6.30, this option can enable the hwptr check using + jiffies. This detects spontaneous invalid pointer callback + values, but can be lead to too much corrections for a (mostly + buggy) hardware that doesn't give smooth pointer updates. + This feature is enabled via the bit 2. + + Bits 3 and 4 are for logging the hwptr records. Note that + these will give flood of kernel messages. + + When bit 5 is set, the driver logs the last 10 xrun errors and + the proc file shows each jiffies, position, period_size, + buffer_size, old_hw_ptr, and hw_ptr_base values. + + When bit 6 is set, the full xrun log is shown only once. + +card*/pcm*/sub*/info + The general information of this PCM sub-stream. + +card*/pcm*/sub*/status + The current status of this PCM sub-stream, elapsed time, + H/W position, etc. + +card*/pcm*/sub*/hw_params + The hardware parameters set for this sub-stream. + +card*/pcm*/sub*/sw_params + The soft parameters set for this sub-stream. + +card*/pcm*/sub*/prealloc + The buffer pre-allocation information. + + +AC97 Codec Information +---------------------- + +card*/codec97#*/ac97#?-? + Shows the general information of this AC97 codec chip, such as + name, capabilities, set up. + +card*/codec97#0/ac97#?-?+regs + Shows the AC97 register dump. Useful for debugging. + + When CONFIG_SND_DEBUG is enabled, you can write to this file for + changing an AC97 register directly. Pass two hex numbers. + For example, + + # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs + + +USB Audio Streams +----------------- + +card*/stream* + Shows the assignment and the current status of each audio stream + of the given card. This information is very useful for debugging. + + +HD-Audio Codecs +--------------- + +card*/codec#* + Shows the general codec information and the attribute of each + widget node. + +card*/eld#* + Available for HDMI or DisplayPort interfaces. + Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, + and describes its audio capabilities and configurations. + + Some ELD fields may be modified by doing `echo name hex_value > eld#*`. + Only do this if you are sure the HDMI sink provided value is wrong. + And if that makes your HDMI audio work, please report to us so that we + can fix it in future kernel releases. + + +Sequencer Information +--------------------- + +seq/drivers + Lists the currently available ALSA sequencer drivers. + +seq/clients + Shows the list of currently available sequencer clients and + ports. The connection status and the running status are shown + in this file, too. + +seq/queues + Lists the currently allocated/running sequencer queues. + +seq/timer + Lists the currently allocated/running sequencer timers. + +seq/oss + Lists the OSS-compatible sequencer stuffs. + + +Help For Debugging? +------------------- + +When the problem is related with PCM, first try to turn on xrun_debug +mode. This will give you the kernel messages when and where xrun +happened. + +If it's really a bug, report it with the following information: + + - the name of the driver/card, show in /proc/asound/cards + - the register dump, if available (e.g. card*/cmipci) + +when it's a PCM problem, + + - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM + sub-stream directory + +when it's a mixer problem, + + - AC97 proc files, codec97#*/* files + +for USB audio/midi, + + - output of lsusb -v + - stream* files in card directory + + +The ALSA bug-tracking system is found at: + + https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44 new file mode 100644 index 00000000..0e41576f --- /dev/null +++ b/Documentation/sound/alsa/README.maya44 @@ -0,0 +1,163 @@ +NOTE: The following is the original document of Rainer's patch that the +current maya44 code based on. Some contents might be obsoleted, but I +keep here as reference -- tiwai + +---------------------------------------------------------------- + +STATE OF DEVELOPMENT: + +This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. +Development is carried out by Rainer Zimmermann (mail@lightshed.de). + +ESI provided a sample Maya44 card for the development work. + +However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing. + +This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008). + + +The following functions work, as tested by Rainer Zimmermann and Piotr Makowski: + +- playback and capture at all sampling rates +- input/output level +- crossmixing +- line/mic switch +- phantom power switch +- analogue monitor a.k.a bypass + + +The following functions *should* work, but are not fully tested: + +- Channel 3+4 analogue - S/PDIF input switching +- S/PDIF output +- all inputs/outputs on the M/IO/DIO extension card +- internal/external clock selection + + +*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.* + + +Things that do not seem to work: + +- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code). + +- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down. + + +DRIVER DETAILS: + +the following files were added: + +pci/ice1724/maya44.c - Maya44 specific code +pci/ice1724/maya44.h +pci/ice1724/ice1724.patch +pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) +i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs +include/wm8776.h + + +Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. +This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately. + + +the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree: + +wtm.h +vt1720_mobo.h +revo.h +prodigy192.h +pontis.h +phase.h +maya44.h +juli.h +aureon.h +amp.h +envy24ht.h +se.h +prodigy_hifi.h + + +*I hope this is the correct way to do things.* + + +SAMPLING RATES: + +The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture. + +As the ICE1724 chip only allows one global sampling rate, this is handled as follows: + +* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels. + +* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices. + +*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality. + + +I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic. + +The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). + + +SOUND DEVICES: + +PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0): + +hw:0,0 input - stereo, analog input 1+2 +hw:0,0 output - stereo, analog output 1+2 +hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input +hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) + + +NAMING OF MIXER CONTROLS: + +(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software). + + +PCM: (digital) output level for channel 1+2 +PCM 1: same for channel 3+4 + +Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. + Make sure this is not turned on while any other source is connected to input 1/2. + It might damage the source and/or the maya44 card. + +Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). + +Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. +Bypass 1: same for channel 3+4. + +Crossmix: cross-mixer from channels 1+2 to channels 3+4 +Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 + +IEC958 Output: switch for S/PDIF output. + This is not supported by the ESI windows driver. + S/PDIF should output the same signal as channel 3+4. [untested!] + + +Digitial output selectors: + + These switches allow a direct digital routing from the ADCs to the DACs. + Each switch determines where the digital input data to one of the DACs comes from. + They are not supported by the ESI windows driver. + For normal operation, they should all be set to "PCM out". + +H/W: Output source channel 1 +H/W 1: Output source channel 2 +H/W 2: Output source channel 3 +H/W 3: Output source channel 4 + +H/W 4 ... H/W 9: unknown function, left in to enable testing. + Possibly some of these control S/PDIF output(s). + If these turn out to be unused, they will go away in later driver versions. + +Selectable values for each of the digital output selectors are: + "PCM out" -> DAC output of the corresponding channel (default setting) + "Input 1"... + "Input 4" -> direct routing from ADC output of the selected input channel + + +-------- + +Feb 14, 2008 +Rainer Zimmermann +mail@lightshed.de + diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt new file mode 100644 index 00000000..f4b5988f --- /dev/null +++ b/Documentation/sound/alsa/SB-Live-mixer.txt @@ -0,0 +1,356 @@ + + Sound Blaster Live mixer / default DSP code + =========================================== + + +The EMU10K1 chips have a DSP part which can be programmed to support +various ways of sample processing, which is described here. +(This article does not deal with the overall functionality of the +EMU10K1 chips. See the manuals section for further details.) + +The ALSA driver programs this portion of chip by default code +(can be altered later) which offers the following functionality: + + +1) IEC958 (S/PDIF) raw PCM +-------------------------- + +This PCM device (it's the 4th PCM device (index 3!) and first subdevice +(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit +little endian streams without any modifications to the digital output +(coaxial or optical). The universal interface allows the creation of up +to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would +be easy to add support for multichannel devices to the current code, +but the conversion routines exist only for stereo (2-channel streams) +at the time. + +Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. + + +2) Digital mixer controls +------------------------- + +These controls are built using the DSP instructions. They offer extended +functionality. Only the default build-in code in the ALSA driver is described +here. Note that the controls work as attenuators: the maximum value is the +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be wrapped +(set to maximal or minimal value without checking of overflow). + + +Explanation of used abbreviations: + +DAC - digital to analog converter +ADC - analog to digital converter +I2S - one-way three wire serial bus for digital sound by Philips Semiconductors + (this standard is used for connecting standalone DAC and ADC converters) +LFE - low frequency effects (subwoofer signal) +AC97 - a chip containing an analog mixer, DAC and ADC converters +IEC958 - S/PDIF +FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. + + +name='Wave Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Wave Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operates +separately (they are not inside the AC97 codec). + +name='Wave Center Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? right DAC PCM slot of the AC97 codec. + +name='Wave LFE Playback Volume',index=0 + +This control is used to attenuate samples for left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is mixed to mono signal (single channel) and forwarded to +the ??rear?? left DAC PCM slot of the AC97 codec. + +name='Wave Capture Volume',index=0 +name='Wave Capture Switch',index=0 + +These controls are used to attenuate samples for left and right PCM FX-bus +accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Synth Playback Volume',index=0 + +This control is used to attenuate samples for left and right MIDI FX-bus +accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result samples are forwarded to the front DAC PCM slots of the AC97 codec. + +name='Synth Capture Volume',index=0 +name='Synth Capture Switch',index=0 + +These controls are used to attenuate samples for left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Surround Playback Volume',index=0 + +This control is used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result samples are forwarded to the rear I2S DACs. These DACs operate +separately (they are not inside the AC97 codec). + +name='Surround Capture Volume',index=0 +name='Surround Capture Switch',index=0 + +These controls are used to attenuate samples for left and right rear PCM FX-bus +accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. +The result is forwarded to the ADC capture FIFO (thus to the standard capture +PCM device). + +name='Center Playback Volume',index=0 + +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? right DAC PCM slot of the AC97 codec. + +name='LFE Playback Volume',index=0 + +This control is used to attenuate sample for center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded +to the ??rear?? left DAC PCM slot of the AC97 codec. + +name='AC97 Playback Volume',index=0 + +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result samples are forwarded to the front DAC PCM +slots of the AC97 codec. +******************************************************************************** +*** Note: This control should be zero for the standard operations, otherwise *** +*** a digital loopback is activated. *** +******************************************************************************** + +name='AC97 Capture Volume',index=0 + +This control is used to attenuate samples for left and right front ADC PCM slots +of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to +the standard capture PCM device). +******************************************************************************** +*** Note: This control should be 100 (maximal value), otherwise no analog *** +*** inputs of the AC97 codec can be captured (recorded). *** +******************************************************************************** + +name='IEC958 TTL Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +name='IEC958 TTL Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 TTL +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='Zoom Video Playback Volume',index=0 + +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the front DAC PCM slots of the AC97 codec. + +name='Zoom Video Capture Volume',index=0 + +This control is used to attenuate samples from left and right zoom video +digital inputs (usually used by a CDROM drive). The result samples are +forwarded to the ADC capture FIFO (thus to the standard capture PCM device). + +name='IEC958 LiveDrive Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital input. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +name='IEC958 LiveDrive Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 optical +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='IEC958 Coaxial Playback Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the front DAC PCM slots +of the AC97 codec. + +name='IEC958 Coaxial Capture Volume',index=0 + +This control is used to attenuate samples from left and right IEC958 coaxial +digital inputs. The result samples are forwarded to the ADC capture FIFO +(thus to the standard capture PCM device). + +name='Line LiveDrive Playback Volume',index=0 +name='Line LiveDrive Playback Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the front +DAC PCM slots of the AC97 codec. + +name='Line LiveDrive Capture Volume',index=1 +name='Line LiveDrive Capture Volume',index=1 + +This control is used to attenuate samples from left and right I2S ADC +inputs (on the LiveDrive). The result samples are forwarded to the ADC +capture FIFO (thus to the standard capture PCM device). + +name='Tone Control - Switch',index=0 + +This control turns the tone control on or off. The samples for front, rear +and center / LFE outputs are affected. + +name='Tone Control - Bass',index=0 + +This control sets the bass intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='Tone Control - Treble',index=0 + +This control sets the treble intensity. There is no neutral value!! +When the tone control code is activated, the samples are always modified. +The closest value to pure signal is 20. + +name='IEC958 Optical Raw Playback Switch',index=0 + +If this switch is on, then the samples for the IEC958 (S/PDIF) digital +output are taken only from the raw FX8010 PCM, otherwise standard front +PCM samples are taken. + +name='Headphone Playback Volume',index=1 + +This control attenuates the samples for the headphone output. + +name='Headphone Center Playback Switch',index=1 + +If this switch is on, then the sample for the center PCM is put to the +left headphone output (useful for SB Live cards without separate center/LFE +output). + +name='Headphone LFE Playback Switch',index=1 + +If this switch is on, then the sample for the center PCM is put to the +right headphone output (useful for SB Live cards without separate center/LFE +output). + + +3) PCM stream related controls +------------------------------ + +name='EMU10K1 PCM Volume',index 0-31 + +Channel volume attenuation in range 0-0xffff. The maximum value (no +attenuation) is default. The channel mapping for three values is +as follows: + + 0 - mono, default 0xffff (no attenuation) + 1 - left, default 0xffff (no attenuation) + 2 - right, default 0xffff (no attenuation) + +name='EMU10K1 PCM Send Routing',index 0-31 + +This control specifies the destination - FX-bus accumulators. There are +twelve values with this mapping: + + 0 - mono, A destination (FX-bus 0-15), default 0 + 1 - mono, B destination (FX-bus 0-15), default 1 + 2 - mono, C destination (FX-bus 0-15), default 2 + 3 - mono, D destination (FX-bus 0-15), default 3 + 4 - left, A destination (FX-bus 0-15), default 0 + 5 - left, B destination (FX-bus 0-15), default 1 + 6 - left, C destination (FX-bus 0-15), default 2 + 7 - left, D destination (FX-bus 0-15), default 3 + 8 - right, A destination (FX-bus 0-15), default 0 + 9 - right, B destination (FX-bus 0-15), default 1 + 10 - right, C destination (FX-bus 0-15), default 2 + 11 - right, D destination (FX-bus 0-15), default 3 + +Don't forget that it's illegal to assign a channel to the same FX-bus accumulator +more than once (it means 0=0 && 1=0 is an invalid combination). + +name='EMU10K1 PCM Send Volume',index 0-31 + +It specifies the attenuation (amount) for given destination in range 0-255. +The channel mapping is following: + + 0 - mono, A destination attn, default 255 (no attenuation) + 1 - mono, B destination attn, default 255 (no attenuation) + 2 - mono, C destination attn, default 0 (mute) + 3 - mono, D destination attn, default 0 (mute) + 4 - left, A destination attn, default 255 (no attenuation) + 5 - left, B destination attn, default 0 (mute) + 6 - left, C destination attn, default 0 (mute) + 7 - left, D destination attn, default 0 (mute) + 8 - right, A destination attn, default 0 (mute) + 9 - right, B destination attn, default 255 (no attenuation) + 10 - right, C destination attn, default 0 (mute) + 11 - right, D destination attn, default 0 (mute) + + + +4) MANUALS/PATENTS: +------------------- + +ftp://opensource.creative.com/pub/doc +------------------------------------- + + Files: + LM4545.pdf AC97 Codec + + m2049.pdf The EMU10K1 Digital Audio Processor + + hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects + + +WIPO Patents +------------ + Patent numbers: + WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) + streams + + WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) + + WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction + Execution and Audio Data Sequencing (Jan. 14, 1999) + + +US Patents (http://www.uspto.gov/) +---------------------------------- + + US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) + + US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) + with a multiport memory onto which multiple asynchronous + digital sound samples can be concurrently loaded + + US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) + + US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) + + US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) + system bus with prioritization and modification of bus transfers + in accordance with loop ends and minimum block sizes + + US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) + pool of short term memory registers + + US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) + a common interrupt by associating programs to GP registers, + defining interrupt register, polling GP registers, and invoking + callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt new file mode 100644 index 00000000..1b0ac06b --- /dev/null +++ b/Documentation/sound/alsa/VIA82xx-mixer.txt @@ -0,0 +1,8 @@ + + VIA82xx mixer + ============= + +On many VIA82xx boards, the 'Input Source Select' mixer control does not work. +Setting it to 'Input2' on such boards will cause recording to hang, or fail +with EIO (input/output error) via OSS emulation. This control should be left +at 'Input1' for such cards. diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt new file mode 100644 index 00000000..0fa40679 --- /dev/null +++ b/Documentation/sound/alsa/alsa-parameters.txt @@ -0,0 +1,135 @@ + ALSA Kernel Parameters + ~~~~~~~~~~~~~~~~~~~~~~ + +See Documentation/kernel-parameters.txt for general information on +specifying module parameters. + +This document may not be entirely up to date and comprehensive. The command +"modinfo -p ${modulename}" shows a current list of all parameters of a loadable +module. Loadable modules, after being loaded into the running kernel, also +reveal their parameters in /sys/module/${modulename}/parameters/. Some of these +parameters may be changed at runtime by the command +"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". + + + snd-ad1816a= [HW,ALSA] + + snd-ad1848= [HW,ALSA] + + snd-ali5451= [HW,ALSA] + + snd-als100= [HW,ALSA] + + snd-als4000= [HW,ALSA] + + snd-azt2320= [HW,ALSA] + + snd-cmi8330= [HW,ALSA] + + snd-cmipci= [HW,ALSA] + + snd-cs4231= [HW,ALSA] + + snd-cs4232= [HW,ALSA] + + snd-cs4236= [HW,ALSA] + + snd-cs4281= [HW,ALSA] + + snd-cs46xx= [HW,ALSA] + + snd-dt019x= [HW,ALSA] + + snd-dummy= [HW,ALSA] + + snd-emu10k1= [HW,ALSA] + + snd-ens1370= [HW,ALSA] + + snd-ens1371= [HW,ALSA] + + snd-es968= [HW,ALSA] + + snd-es1688= [HW,ALSA] + + snd-es18xx= [HW,ALSA] + + snd-es1938= [HW,ALSA] + + snd-es1968= [HW,ALSA] + + snd-fm801= [HW,ALSA] + + snd-gusclassic= [HW,ALSA] + + snd-gusextreme= [HW,ALSA] + + snd-gusmax= [HW,ALSA] + + snd-hdsp= [HW,ALSA] + + snd-ice1712= [HW,ALSA] + + snd-intel8x0= [HW,ALSA] + + snd-interwave= [HW,ALSA] + + snd-interwave-stb= + [HW,ALSA] + + snd-korg1212= [HW,ALSA] + + snd-maestro3= [HW,ALSA] + + snd-mpu401= [HW,ALSA] + + snd-mtpav= [HW,ALSA] + + snd-nm256= [HW,ALSA] + + snd-opl3sa2= [HW,ALSA] + + snd-opti92x-ad1848= + [HW,ALSA] + + snd-opti92x-cs4231= + [HW,ALSA] + + snd-opti93x= [HW,ALSA] + + snd-pmac= [HW,ALSA] + + snd-rme32= [HW,ALSA] + + snd-rme96= [HW,ALSA] + + snd-rme9652= [HW,ALSA] + + snd-sb8= [HW,ALSA] + + snd-sb16= [HW,ALSA] + + snd-sbawe= [HW,ALSA] + + snd-serial= [HW,ALSA] + + snd-sgalaxy= [HW,ALSA] + + snd-sonicvibes= [HW,ALSA] + + snd-sun-amd7930= + [HW,ALSA] + + snd-sun-cs4231= [HW,ALSA] + + snd-trident= [HW,ALSA] + + snd-usb-audio= [HW,ALSA,USB] + + snd-via82xx= [HW,ALSA] + + snd-virmidi= [HW,ALSA] + + snd-wavefront= [HW,ALSA] + + snd-ymfpci= [HW,ALSA] diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt new file mode 100644 index 00000000..751d4503 --- /dev/null +++ b/Documentation/sound/alsa/emu10k1-jack.txt @@ -0,0 +1,74 @@ +This document is a guide to using the emu10k1 based devices with JACK for low +latency, multichannel recording functionality. All of my recent work to allow +Linux users to use the full capabilities of their hardware has been inspired +by the kX Project. Without their work I never would have discovered the true +power of this hardware. + + http://www.kxproject.com + - Lee Revell, 2005.03.30 + +Low latency, multichannel audio with JACK and the emu10k1/emu10k2 +----------------------------------------------------------------- + +Until recently, emu10k1 users on Linux did not have access to the same low +latency, multichannel features offered by the "kX ASIO" feature of their +Windows driver. As of ALSA 1.0.9 this is no more! + +For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback +channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or +even 32 (0.66ms) frames should work well. + +The configuration is slightly more involved than on Windows, as you have to +select the correct device for JACK to use. Actually, for qjackctl users it's +fairly self explanatory - select Duplex, then for capture and playback select +the multichannel devices, set the in and out channels to 16, and the sample +rate to 48000Hz. The command line looks like this: + +/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S + +This will give you 16 input ports and 16 output ports. + +The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the +Audigy). The mapping from FX bus to physical output is described in +SB-Live-mixer.txt (or Audigy-mixer.txt). + +The 16 input ports are connected to the 16 physical inputs. Contrary to +popular belief, all emu10k1 cards are multichannel cards. Which of these +input channels have physical inputs connected to them depends on the card +model. Trial and error is highly recommended; the pinout diagrams +for the card have been reverse engineered by some enterprising kX users and are +available on the internet. Meterbridge is helpful here, and the kX forums are +packed with useful information. + +Each input port will either correspond to a digital (SPDIF) input, an analog +input, or nothing. The one exception is the SBLive! 5.1. On these devices, +the second and third input ports are wired to the center/LFE output. You will +still see 16 capture channels, but only 14 are available for recording inputs. + +This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK +ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output) +channels. + +/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards: +-------------------------------------------- +JACK Epilog FXBUS2(nr) +-------------------------------------------- +capture_1 asio14 FXBUS2(0xe) +capture_2 asio15 FXBUS2(0xf) +capture_3 asio0 FXBUS2(0x0) +~capture_4 Center EXTOUT(0x11) // mapped to by Center +~capture_5 LFE EXTOUT(0x12) // mapped to by LFE +capture_6 asio3 FXBUS2(0x3) +capture_7 asio4 FXBUS2(0x4) +capture_8 asio5 FXBUS2(0x5) +capture_9 asio6 FXBUS2(0x6) +capture_10 asio7 FXBUS2(0x7) +capture_11 asio8 FXBUS2(0x8) +capture_12 asio9 FXBUS2(0x9) +capture_13 asio10 FXBUS2(0xa) +capture_14 asio11 FXBUS2(0xb) +capture_15 asio12 FXBUS2(0xc) +capture_16 asio13 FXBUS2(0xd) +*/ + +TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt new file mode 100644 index 00000000..de8efbc7 --- /dev/null +++ b/Documentation/sound/alsa/hda_codec.txt @@ -0,0 +1,322 @@ +Notes on Universal Interface for Intel High Definition Audio Codec +------------------------------------------------------------------ + +Takashi Iwai + + +[Still a draft version] + + +General +======= + +The snd-hda-codec module supports the generic access function for the +High Definition (HD) audio codecs. It's designed to be independent +from the controller code like ac97 codec module. The real accessors +from/to the controller must be implemented in the lowlevel driver. + +The structure of this module is similar with ac97_codec module. +Each codec chip belongs to a bus class which communicates with the +controller. + + +Initialization of Bus Instance +============================== + +The card driver has to create struct hda_bus at first. The template +struct should be filled and passed to the constructor: + +struct hda_bus_template { + void *private_data; + struct pci_dev *pci; + const char *modelname; + struct hda_bus_ops ops; +}; + +The card driver can set and use the private_data field to retrieve its +own data in callback functions. The pci field is used when the patch +needs to check the PCI subsystem IDs, so on. For non-PCI system, it +doesn't have to be set, of course. +The modelname field specifies the board's specific configuration. The +string is passed to the codec parser, and it depends on the parser how +the string is used. +These fields, private_data, pci and modelname are all optional. + +The ops field contains the callback functions as the following: + +struct hda_bus_ops { + int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int parm); + unsigned int (*get_response)(struct hda_codec *codec); + void (*private_free)(struct hda_bus *); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*pm_notify)(struct hda_codec *codec); +#endif +}; + +The command callback is called when the codec module needs to send a +VERB to the controller. It's always a single command. +The get_response callback is called when the codec requires the answer +for the last command. These two callbacks are mandatory and have to +be given. +The third, private_free callback, is optional. It's called in the +destructor to release any necessary data in the lowlevel driver. + +The pm_notify callback is available only with +CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs +to power up or may power down. The controller should check the all +belonging codecs on the bus whether they are actually powered off +(check codec->power_on), and optionally the driver may power down the +controller side, too. + +The bus instance is created via snd_hda_bus_new(). You need to pass +the card instance, the template, and the pointer to store the +resultant bus instance. + +int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, + struct hda_bus **busp); + +It returns zero if successful. A negative return value means any +error during creation. + + +Creation of Codec Instance +========================== + +Each codec chip on the board is then created on the BUS instance. +To create a codec instance, call snd_hda_codec_new(). + +int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, + struct hda_codec **codecp); + +The first argument is the BUS instance, the second argument is the +address of the codec, and the last one is the pointer to store the +resultant codec instance (can be NULL if not needed). + +The codec is stored in a linked list of bus instance. You can follow +the codec list like: + + struct hda_codec *codec; + list_for_each_entry(codec, &bus->codec_list, list) { + ... + } + +The codec isn't initialized at this stage properly. The +initialization sequence is called when the controls are built later. + + +Codec Access +============ + +To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). +snd_hda_param_read() is for reading parameters. +For writing a sequence of verbs, use snd_hda_sequence_write(). + +There are variants of cached read/write, snd_hda_codec_write_cache(), +snd_hda_sequence_write_cache(). These are used for recording the +register states for the power-management resume. When no PM is needed, +these are equivalent with non-cached version. + +To retrieve the number of sub nodes connected to the given node, use +snd_hda_get_sub_nodes(). The connection list can be obtained via +snd_hda_get_connections() call. + +When an unsolicited event happens, pass the event via +snd_hda_queue_unsol_event() so that the codec routines will process it +later. + + +(Mixer) Controls +================ + +To create mixer controls of all codecs, call +snd_hda_build_controls(). It then builds the mixers and does +initialization stuff on each codec. + + +PCM Stuff +========= + +snd_hda_build_pcms() gives the necessary information to create PCM +streams. When it's called, each codec belonging to the bus stores +codec->num_pcms and codec->pcm_info fields. The num_pcms indicates +the number of elements in pcm_info array. The card driver is supposed +to traverse the codec linked list, read the pcm information in +pcm_info array, and build pcm instances according to them. + +The pcm_info array contains the following record: + +/* PCM information for each substream */ +struct hda_pcm_stream { + unsigned int substreams; /* number of substreams, 0 = not exist */ + unsigned int channels_min; /* min. number of channels */ + unsigned int channels_max; /* max. number of channels */ + hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ + u32 rates; /* supported rates */ + u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ + unsigned int maxbps; /* supported max. bit per sample */ + struct hda_pcm_ops ops; +}; + +/* for PCM creation */ +struct hda_pcm { + char *name; + struct hda_pcm_stream stream[2]; +}; + +The name can be passed to snd_pcm_new(). The stream field contains +the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and +capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver +should pass substreams to snd_pcm_new() for the number of substreams +to create. + +The channels_min, channels_max, rates and formats should be copied to +runtime->hw record. They and maxbps fields are used also to compute +the format value for the HDA codec and controller. Call +snd_hda_calc_stream_format() to get the format value. + +The ops field contains the following callback functions: + +struct hda_pcm_ops { + int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); + int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, + unsigned int stream_tag, unsigned int format, + struct snd_pcm_substream *substream); + int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, + struct snd_pcm_substream *substream); +}; + +All are non-NULL, so you can call them safely without NULL check. + +The open callback should be called in PCM open after runtime->hw is +set up. It may override some setting and constraints additionally. +Similarly, the close callback should be called in the PCM close. + +The prepare callback should be called in PCM prepare. This will set +up the codec chip properly for the operation. The cleanup should be +called in hw_free to clean up the configuration. + +The caller should check the return value, at least for open and +prepare callbacks. When a negative value is returned, some error +occurred. + + +Proc Files +========== + +Each codec dumps the widget node information in +/proc/asound/card*/codec#* file. This information would be really +helpful for debugging. Please provide its contents together with the +bug report. + + +Power Management +================ + +It's simple: +Call snd_hda_suspend() in the PM suspend callback. +Call snd_hda_resume() in the PM resume callback. + + +Codec Preset (Patch) +==================== + +To set up and handle the codec functionality fully, each codec may +have a codec preset (patch). It's defined in struct hda_codec_preset: + + struct hda_codec_preset { + unsigned int id; + unsigned int mask; + unsigned int subs; + unsigned int subs_mask; + unsigned int rev; + const char *name; + int (*patch)(struct hda_codec *codec); + }; + +When the codec id and codec subsystem id match with the given id and +subs fields bitwise (with bitmask mask and subs_mask), the callback +patch is called. The patch callback should initialize the codec and +set the codec->patch_ops field. This is defined as below: + + struct hda_codec_ops { + int (*build_controls)(struct hda_codec *codec); + int (*build_pcms)(struct hda_codec *codec); + int (*init)(struct hda_codec *codec); + void (*free)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + #ifdef CONFIG_PM + int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*resume)(struct hda_codec *codec); + #endif + #ifdef CONFIG_SND_HDA_POWER_SAVE + int (*check_power_status)(struct hda_codec *codec, + hda_nid_t nid); + #endif + }; + +The build_controls callback is called from snd_hda_build_controls(). +Similarly, the build_pcms callback is called from +snd_hda_build_pcms(). The init callback is called after +build_controls to initialize the hardware. +The free callback is called as a destructor. + +The unsol_event callback is called when an unsolicited event is +received. + +The suspend and resume callbacks are for power management. +They can be NULL if no special sequence is required. When the resume +callback is NULL, the driver calls the init callback and resumes the +registers from the cache. If other handling is needed, you'd need to +write your own resume callback. There, the amp values can be resumed +via + void snd_hda_codec_resume_amp(struct hda_codec *codec); +and the other codec registers via + void snd_hda_codec_resume_cache(struct hda_codec *codec); + +The check_power_status callback is called when the amp value of the +given widget NID is changed. The codec code can turn on/off the power +appropriately from this information. + +Each entry can be NULL if not necessary to be called. + + +Generic Parser +============== + +When the device doesn't match with any given presets, the widgets are +parsed via th generic parser (hda_generic.c). Its support is +limited: no multi-channel support, for example. + + +Digital I/O +=========== + +Call snd_hda_create_spdif_out_ctls() from the patch to create controls +related with SPDIF out. + + +Helper Functions +================ + +snd_hda_get_codec_name() stores the codec name on the given string. + +snd_hda_check_board_config() can be used to obtain the configuration +information matching with the device. Define the model string table +and the table with struct snd_pci_quirk entries (zero-terminated), +and pass it to the function. The function checks the modelname given +as a module parameter, and PCI subsystem IDs. If the matching entry +is found, it returns the config field value. + +snd_hda_add_new_ctls() can be used to create and add control entries. +Pass the zero-terminated array of struct snd_kcontrol_new + +Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be +used for the entry of struct snd_kcontrol_new. + +The input MUX helper callbacks for such a control are provided, too: +snd_hda_input_mux_info() and snd_hda_input_mux_put(). See +patch_realtek.c for example. diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt new file mode 100644 index 00000000..7a67ff71 --- /dev/null +++ b/Documentation/sound/alsa/hdspm.txt @@ -0,0 +1,362 @@ +Software Interface ALSA-DSP MADI Driver + +(translated from German, so no good English ;-), +2004 - winfried ritsch + + + + Full functionality has been added to the driver. Since some of + the Controls and startup-options are ALSA-Standard and only the + special Controls are described and discussed below. + + + hardware functionality: + + + Audio transmission: + + number of channels -- depends on transmission mode + + The number of channels chosen is from 1..Nmax. The reason to + use for a lower number of channels is only resource allocation, + since unused DMA channels are disabled and less memory is + allocated. So also the throughput of the PCI system can be + scaled. (Only important for low performance boards). + + Single Speed -- 1..64 channels + + (Note: Choosing the 56channel mode for transmission or as + receiver, only 56 are transmitted/received over the MADI, but + all 64 channels are available for the mixer, so channel count + for the driver) + + Double Speed -- 1..32 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 28 are transmitted/received + over the MADI, but all 32 channels are available for the mixer, + so channel count for the driver + + + Quad Speed -- 1..16 channels + + Note: Choosing the 56-channel mode for + transmission/receive-mode , only 14 are transmitted/received + over the MADI, but all 16 channels are available for the mixer, + so channel count for the driver + + Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) + + Sample Rates -- + + Single Speed -- 32000, 44100, 48000 + + Double Speed -- 64000, 88200, 96000 (untested) + + Quad Speed -- 128000, 176400, 192000 (untested) + + access-mode -- MMAP (memory mapped), Not interleaved + (PCM_NON-INTERLEAVED) + + buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples + + fragments -- 2 + + Hardware-pointer -- 2 Modi + + + The Card supports the readout of the actual Buffer-pointer, + where DMA reads/writes. Since of the bulk mode of PCI it is only + 64 Byte accurate. SO it is not really usable for the + ALSA-mid-level functions (here the buffer-ID gives a better + result), but if MMAP is used by the application. Therefore it + can be configured at load-time with the parameter + precise-pointer. + + + (Hint: Experimenting I found that the pointer is maximum 64 to + large never to small. So if you subtract 64 you always have a + safe pointer for writing, which is used on this mode inside + ALSA. In theory now you can get now a latency as low as 16 + Samples, which is a quarter of the interrupt possibilities.) + + Precise Pointer -- off + interrupt used for pointer-calculation + + Precise Pointer -- on + hardware pointer used. + + Controller: + + + Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to + use the standard mixer-controls, since this would break most of + (especially graphic) ALSA-Mixer GUIs. So Mixer control has be + provided by a 2-dimensional controller using the + hwdep-interface. + + Also all 128+256 Peak and RMS-Meter can be accessed via the + hwdep-interface. Since it could be a performance problem always + copying and converting Peak and RMS-Levels even if you just need + one, I decided to export the hardware structure, so that of + needed some driver-guru can implement a memory-mapping of mixer + or peak-meters over ioctl, or also to do only copying and no + conversion. A test-application shows the usage of the controller. + + Latency Controls --- not implemented !!! + + + Note: Within the windows-driver the latency is accessible of a + control-panel, but buffer-sizes are controlled with ALSA from + hwparams-calls and should not be changed in run-state, I did not + implement it here. + + + System Clock -- suspended !!!! + + Name -- "System Clock Mode" + + Access -- Read Write + + Values -- "Master" "Slave" + + + !!!! This is a hardware-function but is in conflict with the + Clock-source controller, which is a kind of ALSA-standard. I + makes sense to set the card to a special mode (master at some + frequency or slave), since even not using an Audio-application + a studio should have working synchronisations setup. So use + Clock-source-controller instead !!!! + + Clock Source + + Name -- "Sample Clock Source" + + Access -- Read Write + + Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", + "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", + "Internal 96.0 kHz" + + Choose between Master at a specific Frequency and so also the + Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref" + + + !!!! This is no pure hardware function but was implemented by + ALSA by some ALSA-drivers before, so I use it also. !!! + + + Preferred Sync Ref + + Name -- "Preferred Sync Reference" + + Access -- Read Write + + Values -- "Word" "MADI" + + + Within the Auto-sync-Mode the preferred Sync Source can be + chosen. If it is not available another is used if possible. + + Note: Since MADI has a much higher bit-rate than word-clock, the + card should synchronise better in MADI Mode. But since the + RME-PLL is very good, there are almost no problems with + word-clock too. I never found a difference. + + + TX 64 channel --- + + Name -- "TX 64 channels mode" + + Access -- Read Write + + Values -- 0 1 + + Using 64-channel-modus (1) or 56-channel-modus for + MADI-transmission (0). + + + Note: This control is for output only. Input-mode is detected + automatically from hardware sending MADI. + + + Clear TMS --- + + Name -- "Clear Track Marker" + + Access -- Read Write + + Values -- 0 1 + + + Don't use to lower 5 Audio-bits on AES as additional Bits. + + + Safe Mode oder Auto Input --- + + Name -- "Safe Mode" + + Access -- Read Write + + Values -- 0 1 + + (default on) + + If on (1), then if either the optical or coaxial connection + has a failure, there is a takeover to the working one, with no + sample failure. Its only useful if you use the second as a + backup connection. + + Input --- + + Name -- "Input Select" + + Access -- Read Write + + Values -- optical coaxial + + + Choosing the Input, optical or coaxial. If Safe-mode is active, + this is the preferred Input. + +-------------- Mixer ---------------------- + + Mixer + + Name -- "Mixer" + + Access -- Read Write + + Values - + + + Here as a first value the channel-index is taken to get/set the + corresponding mixer channel, where 0-63 are the input to output + fader and 64-127 the playback to outputs fader. Value 0 + is channel muted 0 and 32768 an amplification of 1. + + Chn 1-64 + + fast mixer for the ALSA-mixer utils. The diagonal of the + mixer-matrix is implemented from playback to output. + + + Line Out + + Name -- "Line Out" + + Access -- Read Write + + Values -- 0 1 + + Switching on and off the analog out, which has nothing to do + with mixing or routing. the analog outs reflects channel 63,64. + + +--- information (only read access): + + Sample Rate + + Name -- "System Sample Rate" + + Access -- Read-only + + getting the sample rate. + + + External Rate measured + + Name -- "External Rate" + + Access -- Read only + + + Should be "Autosync Rate", but Name used is + ALSA-Scheme. External Sample frequency liked used on Autosync is + reported. + + + MADI Sync Status + + Name -- "MADI Sync Lock Status" + + Access -- Read + + Values -- 0,1,2 + + MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. + + + Word Clock Sync Status + + Name -- "Word Clock Lock Status" + + Access -- Read + + Values -- 0,1,2 + + Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. + + AutoSync + + Name -- "AutoSync Reference" + + Access -- Read + + Values -- "WordClock", "MADI", "None" + + Sync-Reference is either "WordClock", "MADI" or none. + + RX 64ch --- noch nicht implementiert + + MADI-Receiver is in 64 channel mode oder 56 channel mode. + + + AB_inp --- not tested + + Used input for Auto-Input. + + + actual Buffer Position --- not implemented + + !!! this is a ALSA internal function, so no control is used !!! + + + +Calling Parameter: + + index int array (min = 1, max = 8), + "Index value for RME HDSPM interface." card-index within ALSA + + note: ALSA-standard + + id string array (min = 1, max = 8), + "ID string for RME HDSPM interface." + + note: ALSA-standard + + enable int array (min = 1, max = 8), + "Enable/disable specific HDSPM sound-cards." + + note: ALSA-standard + + precise_ptr int array (min = 1, max = 8), + "Enable precise pointer, or disable." + + note: Use only when the application supports this (which is a special case). + + line_outs_monitor int array (min = 1, max = 8), + "Send playback streams to analog outs by default." + + + note: each playback channel is mixed to the same numbered output + channel (routed). This is against the ALSA-convention, where all + channels have to be muted on after loading the driver, but was + used before on other cards, so i historically use it again) + + + + enable_monitor int array (min = 1, max = 8), + "Enable Analog Out on Channel 63/64 by default." + + note: here the analog output is enabled (but not routed). \ No newline at end of file diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt new file mode 100644 index 00000000..9657e809 --- /dev/null +++ b/Documentation/sound/alsa/powersave.txt @@ -0,0 +1,41 @@ +Notes on Power-Saving Mode +========================== + +AC97 and HD-audio drivers have the automatic power-saving mode. +This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE +and CONFIG_SND_HDA_POWER_SAVE options, respectively. + +With the automatic power-saving, the driver turns off the codec power +appropriately when no operation is required. When no applications use +the device and/or no analog loopback is set, the power disablement is +done fully or partially. It'll save a certain power consumption, thus +good for laptops (even for desktops). + +The time-out for automatic power-off can be specified via power_save +module option of snd-ac97-codec and snd-hda-intel modules. Specify +the time-out value in seconds. 0 means to disable the automatic +power-saving. The default value of timeout is given via +CONFIG_SND_AC97_POWER_SAVE_DEFAULT and +CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 +(the minimum value) isn't recommended because many applications try to +reopen the device frequently. 10 would be a good choice for normal +operations. + +The power_save option is exported as writable. This means you can +adjust the value via sysfs on the fly. For example, to turn on the +automatic power-save mode with 10 seconds, write to +/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): + + # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + + +Note that you might hear click noise/pop when changing the power +state. Also, it often takes certain time to wake up from the +power-down to the active state. These are often hardly to fix, so +don't report extra bug reports unless you have a fix patch ;-) + +For HD-audio interface, there is another module option, +power_save_controller. This enables/disables the power-save mode of +the controller side. Setting this on may reduce a bit more power +consumption, but might result in longer wake-up time and click noise. +Try to turn it off when you experience such a thing too often. diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html new file mode 100644 index 00000000..d9776cf6 --- /dev/null +++ b/Documentation/sound/alsa/seq_oss.html @@ -0,0 +1,409 @@ + + + + OSS Sequencer Emulation on ALSA + + + +
+

+ +

+ +
+

+OSS Sequencer Emulation on ALSA

+ +
+

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

ver.0.1.8; Nov. 16, 1999 +

+ +

+ +

+1. Description

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

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

The following features are emulated by this driver: +

    +
  • +Normal sequencer and MIDI events:
    • + +
    They are converted to the ALSA sequencer events, and sent to the corresponding +port. +
  • +Timer events:
    • + +
    The timer is not selectable by ioctl. The control rate is fixed to +100 regardless of HZ. That is, even on Alpha system, a tick is always +1/100 second. The base rate and tempo can be changed in /dev/music. + +
  • +Patch loading:
    • + +
    It purely depends on the synth drivers whether it's supported since +the patch loading is realized by callback to the synth driver. +
  • +I/O controls:
    • + +
    Most of controls are accepted. Some controls +are dependent on the synth driver, as well as even on original OSS.
+Furthermore, you can find the following advanced features: +
    +
  • +Better queue mechanism:
    • + +
    The events are queued before processing them. +
  • +Multiple applications:
    • + +
    You can run two or more applications simultaneously (even for OSS sequencer)! +However, each MIDI device is exclusive - that is, if a MIDI device is opened +once by some application, other applications can't use it. No such a restriction +in synth devices. +
  • +Real-time event processing:
    • + +
    The events can be processed in real time without using out of bound +ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed +events will be processed in real-time without queued. To switch off the +real-time mode, send RELTIME 0 event. +
  • +/proc interface:
    • + +
    The status of applications and devices can be shown via /proc/asound/seq/oss +at any time. In the later version, configuration will be changed via /proc +interface, too.
+ +

+2. Installation

+Run configure script with both sequencer support (--with-sequencer=yes) +and OSS emulation (--with-oss=yes) options. A module snd-seq-oss.o +will be created. If the synth module of your sound card supports for OSS +emulation (so far, only Emu8000 driver), this module will be loaded automatically. +Otherwise, you need to load this module manually. +

At beginning, this module probes all the MIDI ports which have been +already connected to the sequencer. Once after that, the creation and deletion +of ports are watched by announcement mechanism of ALSA sequencer. +

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

        OSS sequencer emulation version 0.1.8
+        ALSA client number 63
+        ALSA receiver port 0
+
+        Number of applications: 0
+
+        Number of synth devices: 1
+
+        synth 0: [EMU8000]
+          type 0x1 : subtype 0x20 : voices 32
+          capabilties : ioctl enabled / load_patch enabled
+
+        Number of MIDI devices: 3
+
+        midi 0: [Emu8000 Port-0] ALSA port 65:0
+          capability write / opened none
+
+        midi 1: [Emu8000 Port-1] ALSA port 65:1
+          capability write / opened none
+
+        midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+          capability read/write / opened none
+Note that the device number may be different from the information of +/proc/asound/oss-devices +or ones of the original OSS driver. Use the device number listed in /proc/asound/seq/oss +to play via OSS sequencer emulation. +

+3. Using Synthesizer Devices

+Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 +and xmp-1.1.5. You can load samples via /dev/sequencer like sfxload, +too. +

If the lowlevel driver supports multiple access to synth devices (like +Emu8000 driver), two or more applications are allowed to run at the same +time. +

+4. Using MIDI Devices

+So far, only MIDI output was tested. MIDI input was not checked at all, +but hopefully it will work. Use the device number listed in /proc/asound/seq/oss. +Be aware that these numbers are mostly different from the list in +/proc/asound/oss-devices. +

+5. Module Options

+The following module options are available: +
    +
  • +maxqlen
    • + +
    specifies the maximum read/write queue length. This queue is private +for OSS sequencer, so that it is independent from the queue length of ALSA +sequencer. Default value is 1024. +
  • +seq_oss_debug
    • + +
    specifies the debug level and accepts zero (= no debug message) or +positive integer. Default value is 0.
+ +

+6. Queue Mechanism

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

All the events from /dev/sequencer are parsed at beginning. +The timing events are also parsed at this moment, so that the events may +be processed in real-time. Sending an event ABSTIME 0 switches the operation +mode to real-time mode, and sending an event RELTIME 0 switches it off. +In the real-time mode, all events are dispatched immediately. +

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

If the write-queue is full, the application sleeps until a certain amount +(as default one half) becomes empty in blocking mode. The synchronization +to write timing was implemented, too. +

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

+7. Interface to Synthesizer Device

+ +

+7.1. Registration

+To register an OSS synthesizer device, use snd_seq_oss_synth_register +function. +
int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+                              snd_seq_oss_callback_t *oper, void *private_data)
+The arguments name, type, subtype and +nvoices +are used for making the appropriate synth_info structure for ioctl. The +return value is an index number of this device. This index must be remembered +for unregister. If registration is failed, -errno will be returned. +

To release this device, call snd_seq_oss_synth_unregister function: +

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

+7.2. Callbacks

+OSS synthesizer devices have capability for sample downloading and ioctls +like sample reset. In OSS emulation, these special features are realized +by using callbacks. The registration argument oper is used to specify these +callbacks. The following callback functions must be defined: +
snd_seq_oss_callback_t:
+        int (*open)(snd_seq_oss_arg_t *p, void *closure);
+        int (*close)(snd_seq_oss_arg_t *p);
+        int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+        int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+        int (*reset)(snd_seq_oss_arg_t *p);
+Except for open and close callbacks, they are allowed
+to be NULL.
+
Each callback function takes the argument type snd_seq_oss_arg_t as the
+first argument.
+
struct snd_seq_oss_arg_t {
+        int app_index;
+        int file_mode;
+        int seq_mode;
+        snd_seq_addr_t addr;
+        void *private_data;
+        int event_passing;
+};

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

+7.2.1. Open Callback

+The open is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+
    
+
  1. 
+Allocate private data record.
    2. 
+
+
  2. 
+Create an ALSA sequencer port.
    3. 
+
+
  3. 
+Set the new port address on arg->addr.
    4. 
+
+
  4. 
+Set the private data record pointer on arg->private_data.
    5. 
+

+Note that the type bit-flags in port_info of this synth port must NOT contain
+TYPE_MIDI_GENERIC
+bit. Instead, TYPE_SPECIFIC should be used. Also, CAP_SUBSCRIPTION
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+

+7.2.2 Ioctl Callback

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

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

+7.2.3 Load_Patch Callback

+The load_patch callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+

+7.2.4 Close Callback

+The close callback is called when this device is closed by the
+applicaion. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+

+7.2.5 Reset Callback

+The reset callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a HEARTBEAT event to the
+port.
+

+7.3 Events

+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be implemented
+by the driver:
+
 
+
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ALSA eventOriginal OSS events
NOTEONSEQ_NOTEON
+
MIDI_NOTEON
NOTESEQ_NOTEOFF
+
MIDI_NOTEOFF
KEYPRESSMIDI_KEY_PRESSURE
CHANPRESSSEQ_AFTERTOUCH
+
MIDI_CHN_PRESSURE
PGMCHANGESEQ_PGMCHANGE
+
MIDI_PGM_CHANGE
PITCHBENDSEQ_CONTROLLER(CTRL_PITCH_BENDER)
+
MIDI_PITCH_BEND
CONTROLLERMIDI_CTL_CHANGE
+
SEQ_BALANCE (with CTL_PAN)
CONTROL14SEQ_CONTROLLER
REGPARAMSEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
SYSEXSEQ_SYSEX

+
+
The most of these behavior can be realized by MIDI emulation driver
+included in the Emu8000 lowlevel driver. In the future release, this module
+will be independent.
+
Some OSS events (SEQ_PRIVATE and SEQ_VOLUME events) are passed as event
+type SND_SEQ_OSS_PRIVATE.  The OSS sequencer passes these event 8 byte
+packets without any modification. The lowlevel driver should process these
+events appropriately.
+

+8. Interface to MIDI Device

+Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer
+ports automatically by receiving announcement from ALSA sequencer, the
+MIDI devices don't need to be registered explicitly like synth devices.
+However, the MIDI port_info registered to ALSA sequencer must include a group
+name SND_SEQ_GROUP_DEVICE and a capability-bit CAP_READ or
+CAP_WRITE. Also, subscription capabilities, CAP_SUBS_READ or CAP_SUBS_WRITE,
+must be defined, too. If these conditions are not satisfied, the port is not
+registered as OSS sequencer MIDI device.
+
The events via MIDI devices are parsed in OSS sequencer and converted
+to the corresponding ALSA sequencer events. The input from MIDI sequencer
+is also converted to MIDI byte events by OSS sequencer. This works just
+a reverse way of seq_midi module.
+

+9. Known Problems / TODO's

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

+
+
+
diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt
new file mode 100644
index 00000000..c1919559
--- /dev/null
+++ b/Documentation/sound/alsa/serial-u16550.txt
@@ -0,0 +1,88 @@
+
+			Serial UART 16450/16550 MIDI driver
+			===================================
+
+The adaptor module parameter allows you to select either:
+
+  0 - Roland Soundcanvas support (default)
+  1 - Midiator MS-124T support (1)
+  2 - Midiator MS-124W S/A mode (2)
+  3 - MS-124W M/B mode support (3)
+  4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15).  Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1.  Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors.  Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+			speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16.  The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4.  Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports.  M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs. 
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported. 
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with.  The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port.  Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow.  The number of
+inputs is specified by the ins parameter.
diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
index 00000000..c9679264
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DAI.txt
@@ -0,0 +1,56 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+  AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at :-
+http://www.intel.com/p/en_US/business/design
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
+         transition.
+
+ o Left Justified - MSB is transmitted on transition of LRC.
+
+ o Right Justified - MSB is transmitted sample size BCLKs before LRC
+                     transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, which can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus simultaneously (this
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
new file mode 100644
index 00000000..b1300162
--- /dev/null
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -0,0 +1,51 @@
+Audio Clocking
+==============
+
+This text describes the audio clocking terms in ASoC and digital audio in
+general. Note: Audio clocking can be complex!
+
+
+Master Clock
+------------
+
+Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
+or SYSCLK). This audio master clock can be derived from a number of sources
+(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
+audio playback and capture sample rates.
+
+Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at a set frequency (i.e. crystals).
+
+
+DAI Clocks
+----------
+The Digital Audio Interface is usually driven by a Bit Clock (often referred to
+as BCLK). This clock is used to drive the digital audio data across the link
+between the codec and CPU.
+
+The DAI also has a frame clock to signal the start of each audio frame. This
+clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
+runs at exactly the sample rate (LRC = Rate).
+
+Bit Clock can be generated as follows:-
+
+BCLK = MCLK / x
+
+ or
+
+BCLK = LRC * x
+
+ or
+
+BCLK = LRC * Channels * Word Size
+
+This relationship depends on the codec or SoC CPU in particular. In general
+it is best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and word size) to save on power.
+
+It is also desirable to use the codec (if possible) to drive (or master) the
+audio clocks as it usually gives more accurate sample rates than the CPU.
+
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 00000000..bce23a4a
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,195 @@
+ASoC Codec Driver
+=================
+
+The codec driver is generic and hardware independent code that configures the
+codec to provide audio capture and playback. It should contain no code that is
+specific to the target platform or machine. All platform and machine specific
+code should be added to the platform and machine drivers respectively.
+
+Each codec driver *must* provide the following features:-
+
+ 1) Codec DAI and PCM configuration
+ 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 3) Mixers and audio controls
+ 4) Codec audio operations
+
+Optionally, codec drivers can also provide:-
+
+ 5) DAPM description.
+ 6) DAPM event handler.
+ 7) DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Codec DAI and PCM configuration
+-----------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+	.name = "wm8731-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.ops = &wm8731_dai_ops,
+	.symmetric_rates = 1,
+};
+
+
+2 - Codec control IO
+--------------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec drivers provide
+functions to read and write the codec registers along with supplying a
+register cache:-
+
+	/* IO control data and register cache */
+	void *control_data; /* codec control (i2c/3wire) data */
+	void *reg_cache;
+
+Codec read/write should do any data formatting and call the hardware
+read write below to perform the IO. These functions are called by the
+core and ALSA when performing DAPM or changing the mixer:-
+
+    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
+
+Codec hardware IO functions - usually points to either the I2C, SPI or AC97
+read/write:-
+
+	hw_write_t hw_write;
+	hw_read_t hw_read;
+
+
+3 - Mixers and audio controls
+-----------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+
+    #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+
+  xname = Control name e.g. "Playback Volume"
+  reg = codec register
+  shift = control bit(s) offset in register
+  mask = control bit size(s) e.g. mask of 7 = 3 bits
+  invert = the control is inverted
+
+Other macros include:-
+
+    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+
+    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+
+    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+
+   xreg = register
+   xshift = control bit(s) offset in register
+   xmask = control bit(s) size
+   xtexts = pointer to array of strings that describe each setting
+
+   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+4 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+};
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+5 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+6 - DAPM event handler
+----------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+
+	SNDRV_CTL_POWER_D0: /* full On */
+	/* vref/mid, clk and osc on, active */
+
+	SNDRV_CTL_POWER_D1: /* partial On */
+	SNDRV_CTL_POWER_D2: /* partial On */
+
+	SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	/* everything off except vref/vmid, inactive */
+
+	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+7 - Codec DAC digital mute control
+----------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise.  The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		snd_soc_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+}
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
new file mode 100644
index 00000000..05bf5a0e
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,294 @@
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+1. Description
+==============
+
+Dynamic Audio Power Management (DAPM) is designed to allow portable
+Linux devices to use the minimum amount of power within the audio
+subsystem at all times. It is independent of other kernel PM and as
+such, can easily co-exist with the other PM systems.
+
+DAPM is also completely transparent to all user space applications as
+all power switching is done within the ASoC core. No code changes or
+recompiling are required for user space applications. DAPM makes power
+switching decisions based upon any audio stream (capture/playback)
+activity and audio mixer settings within the device.
+
+DAPM spans the whole machine. It covers power control within the entire
+audio subsystem, this includes internal codec power blocks and machine
+level power systems.
+
+There are 4 power domains within DAPM
+
+   1. Codec domain - VREF, VMID (core codec and audio power)
+      Usually controlled at codec probe/remove and suspend/resume, although
+      can be set at stream time if power is not needed for sidetone, etc.
+
+   2. Platform/Machine domain - physically connected inputs and outputs
+      Is platform/machine and user action specific, is configured by the
+      machine driver and responds to asynchronous events e.g when HP
+      are inserted
+
+   3. Path domain - audio susbsystem signal paths
+      Automatically set when mixer and mux settings are changed by the user.
+      e.g. alsamixer, amixer.
+
+   4. Stream domain - DACs and ADCs.
+      Enabled and disabled when stream playback/capture is started and
+      stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching decisions are made automatically by consulting an audio
+routing map of the whole machine. This map is specific to each machine and
+consists of the interconnections between every audio component (including
+internal codec components). All audio components that effect power are called
+widgets hereafter.
+
+
+2. DAPM Widgets
+===============
+
+Audio DAPM widgets fall into a number of types:-
+
+ o Mixer      - Mixes several analog signals into a single analog signal.
+ o Mux        - An analog switch that outputs only one of many inputs.
+ o PGA        - A programmable gain amplifier or attenuation widget.
+ o ADC        - Analog to Digital Converter
+ o DAC        - Digital to Analog Converter
+ o Switch     - An analog switch
+ o Input      - A codec input pin
+ o Output     - A codec output pin
+ o Headphone  - Headphone (and optional Jack)
+ o Mic        - Mic (and optional Jack)
+ o Line       - Line Input/Output (and optional Jack)
+ o Speaker    - Speaker
+ o Supply     - Power or clock supply widget used by other widgets.
+ o Pre        - Special PRE widget (exec before all others)
+ o Post       - Special POST widget (exec after all others)
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets are usually added in the codec driver and the machine driver. There are
+convenience macros defined in soc-dapm.h that can be used to quickly build a
+list of widgets of the codecs and machines DAPM widgets.
+
+Most widgets have a name, register, shift and invert. Some widgets have extra
+parameters for stream name and kcontrols.
+
+
+2.1 Stream Domain Widgets
+-------------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADCs
+(analog to digital converters) and DACs (digital to analog converters).
+
+Stream widgets have the following format:-
+
+SND_SOC_DAPM_DAC(name, stream name, 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),
+
+
+2.2 Path Domain Widgets
+-----------------------
+
+Path domain widgets have a ability to control or affect the audio signal or
+audio paths within the audio subsystem. They have the following form:-
+
+SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
+
+Any widget kcontrols can be set using the controls and num_controls members.
+
+e.g. Mixer widget (the kcontrols are declared first)
+
+/* Output Mixer */
+static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
+	ARRAY_SIZE(wm8731_output_mixer_controls)),
+
+If you dont want the mixer elements prefixed with the name of the mixer widget,
+you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
+as for SND_SOC_DAPM_MIXER.
+
+2.3 Platform/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) that can be independently powered. e.g.
+
+ o Speaker Amp
+ o Microphone Bias
+ o Jack connectors
+
+A machine widget can have an optional call back.
+
+e.g. Jack connector widget for an external Mic that enables Mic Bias
+when the Mic is inserted:-
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
+{
+	gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+
+
+2.4 Codec Domain
+----------------
+
+The codec power domain has no widgets and is handled by the codecs DAPM event
+handler. This handler is called when the codec powerstate is changed wrt to any
+stream event or by kernel PM events.
+
+
+2.5 Virtual Widgets
+-------------------
+
+Sometimes widgets exist in the codec or machine audio map that don't have any
+corresponding soft power control. In this case it is necessary to create
+a virtual widget - a widget with no control bits e.g.
+
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
+
+This can be used to merge to signal paths together in software.
+
+After all the widgets have been defined, they can then be added to the DAPM
+subsystem individually with a call to snd_soc_dapm_new_control().
+
+
+3. Codec Widget Interconnections
+================================
+
+Widgets are connected to each other within the codec and machine by audio paths
+(called interconnections). Each interconnection must be defined in order to
+create a map of all audio paths between widgets.
+
+This is easiest with a diagram of the codec (and schematic of the machine audio
+system), as it requires joining widgets together via their audio signal paths.
+
+e.g., from the WM8731 output mixer (wm8731.c)
+
+The WM8731 output mixer has 3 inputs (sources)
+
+ 1. Line Bypass Input
+ 2. DAC (HiFi playback)
+ 3. Mic Sidetone Input
+
+Each input in this example has a kcontrol associated with it (defined in example
+above) and is connected to the output mixer via its kcontrol name. We can now
+connect the destination widget (wrt audio signal) with its source widgets.
+
+	/* output mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+So we have :-
+
+	Destination Widget  <=== Path Name <=== Source Widget
+
+Or:-
+
+	Sink, Path, Source
+
+Or :-
+
+	"Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
+
+When there is no path name connecting widgets (e.g. a direct connection) we
+pass NULL for the path name.
+
+Interconnections are created with a call to:-
+
+snd_soc_dapm_connect_input(codec, sink, path, source);
+
+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+interconnections have been registered with the core. This causes the core to
+scan the codec and machine so that the internal DAPM state matches the
+physical state of the machine.
+
+
+3.1 Machine Widget Interconnections
+-----------------------------------
+Machine widget interconnections are created in the same way as codec ones and
+directly connect the codec pins to machine level widgets.
+
+e.g. connects the speaker out codec pins to the internal speaker.
+
+	/* ext speaker connected to codec pins LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+This allows the DAPM to power on and off pins that are connected (and in use)
+and pins that are NC respectively.
+
+
+4 Endpoint Widgets
+===================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+ o Headphone Jack
+ o Internal Speaker
+ o Internal Mic
+ o Mic Jack
+ o Codec Pins
+
+When a codec pin is NC it can be marked as not used with a call to
+
+snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
+
+The last argument is 0 for inactive and 1 for active. This way the pin and its
+input widget will never be powered up and consume power.
+
+This also applies to machine widgets. e.g. if a headphone is connected to a
+jack then the jack can be marked active. If the headphone is removed, then
+the headphone jack can be marked inactive.
+
+
+5 DAPM Widget Events
+====================
+
+Some widgets can register their interest with the DAPM core in PM events.
+e.g. A Speaker with an amplifier registers a widget so the amplifier can be
+powered only when the spk is in use.
+
+/* turn speaker amplifier on/off depending on use */
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
+	SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+
+Please see soc-dapm.h for all other widgets that support events.
+
+
+5.1 Event types
+---------------
+
+The following event types are supported by event widgets.
+
+/* dapm event types */
+#define SND_SOC_DAPM_PRE_PMU	0x1 	/* before widget power up */
+#define SND_SOC_DAPM_POST_PMU	0x2		/* after widget power up */
+#define SND_SOC_DAPM_PRE_PMD	0x4 	/* before widget power down */
+#define SND_SOC_DAPM_POST_PMD	0x8		/* after widget power down */
+#define SND_SOC_DAPM_PRE_REG	0x10	/* before audio path setup */
+#define SND_SOC_DAPM_POST_REG	0x20	/* after audio path setup */
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 00000000..fcf82a41
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h.  ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+   user visible jack.  In embedded systems it is common for multiple
+   to be present on a single jack but handled by separate bits of
+   hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+   automatically based on the detected jack status (eg, turning off the
+   headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone.  Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space.  The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack.  Each snd_soc_jack has zero or more of these
+which are updated automatically.  They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins().  The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update.  The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function.  Other methods are
+also available, for example integrated into CODECs.  One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware.  The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 00000000..3e2ec9cb
--- /dev/null
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -0,0 +1,93 @@
+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together the platform
+and codec drivers.
+
+The machine driver can contain codec and platform specific code. It registers
+the audio subsystem with the kernel as a platform device and is represented by
+the following struct:-
+
+/* SoC machine */
+struct snd_soc_card {
+	char *name;
+
+	...
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+
+	/* the pre and post PM functions are used to do any PM work before and
+	 * after the codec and DAIs do any PM work. */
+	int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
+	int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
+	int (*resume_pre)(struct platform_device *pdev);
+	int (*resume_post)(struct platform_device *pdev);
+
+	...
+
+	/* CPU <--> Codec DAI links  */
+	struct snd_soc_dai_link *dai_link;
+	int num_links;
+
+	...
+};
+
+probe()/remove()
+----------------
+probe/remove are optional. Do any machine specific probe here.
+
+
+suspend()/resume()
+------------------
+The machine driver has pre and post versions of suspend and resume to take care
+of any machine audio tasks that have to be done before or after the codec, DAIs
+and DMA is suspended and resumed. Optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAIs together. It can
+also be used to set up the DAI system clock and for any machine related DAI
+initialisation e.g. the machine audio map can be connected to the codec audio
+map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
+for examples.
+
+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai_name = "pxa-is2-dai",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8713-codec.0-001a",
+	.init = corgi_wm8731_init,
+	.ops = &corgi_ops,
+};
+
+struct snd_soc_card then sets up the machine with its DAIs. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_card snd_soc_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+};
+
+
+Machine Power Map
+-----------------
+
+The machine driver can optionally extend the codec power map and to become an
+audio power map of the audio subsystem. This allows for automatic power up/down
+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
+sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
+details.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the DAI init function.
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 00000000..138ac88c
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,86 @@
+ALSA SoC Layer
+==============
+
+The overall project goal of the ALSA System on Chip (ASoC) layer is to
+provide better ALSA support for embedded system-on-chip processors (e.g.
+pxa2xx, au1x00, iMX, etc) and portable audio codecs.  Prior to the ASoC
+subsystem there was some support in the kernel for SoC audio, however it
+had some limitations:-
+
+  * Codec drivers were often tightly coupled to the underlying SoC
+    CPU. This is not ideal and leads to code duplication - for example,
+    Linux had different wm8731 drivers for 4 different SoC platforms.
+
+  * There was no standard method to signal user initiated audio events (e.g.
+    Headphone/Mic insertion, Headphone/Mic detection after an insertion
+    event). These are quite common events on portable devices and often require
+    machine specific code to re-route audio, enable amps, etc., after such an
+    event.
+
+  * Drivers tended to power up the entire codec when playing (or
+    recording) audio. This is fine for a PC, but tends to waste a lot of
+    power on portable devices. There was also no support for saving
+    power via changing codec oversampling rates, bias currents, etc.
+
+
+ASoC Design
+===========
+
+The ASoC layer is designed to address these issues and provide the following
+features :-
+
+  * Codec independence. Allows reuse of codec drivers on other platforms
+    and machines.
+
+  * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
+    interface and codec registers its audio interface capabilities with the
+    core and are subsequently matched and configured when the application
+    hardware parameters are known.
+
+  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+    its minimum power state at all times. This includes powering up/down
+    internal power blocks depending on the internal codec audio routing and any
+    active streams.
+
+  * Pop and click reduction. Pops and clicks can be reduced by powering the
+    codec up/down in the correct sequence (including using digital mute). ASoC
+    signals the codec when to change power states.
+
+  * Machine specific controls: Allow machines to add controls to the sound card
+    (e.g. volume control for speaker amplifier).
+
+To achieve all this, ASoC basically splits an embedded audio system into 3
+components :-
+
+  * Codec driver: The codec driver is platform independent and contains audio
+    controls, audio interface capabilities, codec DAPM definition and codec IO
+    functions.
+
+  * Platform driver: The platform driver contains the audio DMA engine and audio
+    interface drivers (e.g. I2S, AC97, PCM) for that platform.
+
+  * Machine driver: The machine driver handles any machine specific controls and
+    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.
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 00000000..d57efad3
--- /dev/null
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -0,0 +1,66 @@
+ASoC Platform Driver
+====================
+
+An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
+and control. The platform drivers only target the SoC CPU and must have no board
+specific code.
+
+Audio DMA
+=========
+
+The platform DMA driver optionally supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+	int (*trigger)(struct snd_pcm_substream *, int);
+};
+
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
+
+struct snd_soc_platform_driver {
+	char *name;
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+	int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+	int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
+	void (*pcm_free)(struct snd_pcm *);
+
+	/*
+	 * For platform caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+
+	/* platform stream ops */
+	struct snd_pcm_ops *pcm_ops;
+};
+
+Please refer to the ALSA driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+An example DMA driver is soc/pxa/pxa2xx-pcm.c
+
+
+SoC DAI Drivers
+===============
+
+Each SoC DAI driver must provide the following features:-
+
+ 1) Digital audio interface (DAI) description
+ 2) Digital audio interface configuration
+ 3) PCM's description
+ 4) SYSCLK configuration
+ 5) Suspend and resume (optional)
+
+Please see codec.txt for a description of items 1 - 4.
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
new file mode 100644
index 00000000..e1e74daa
--- /dev/null
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is noticeable on PCs when an
+audio module is either loaded or unloaded (at module load time the sound card is
+powered up and causes a popping noise on the speakers).
+
+Pops and clicks can be more frequent on portable systems with DAPM. This is
+because the components within the subsystem are being dynamically powered
+depending on the audio usage and this can subsequently cause a small pop or
+click every time a component power state is changed.
+
+
+Minimising Playback Pops and Clicks
+===================================
+
+Playback pops in portable audio subsystems cannot be completely eliminated
+currently, however future audio codec hardware will have better pop and click
+suppression.  Pops can be reduced within playback by powering the audio
+components in a specific order. This order is different for startup and
+shutdown and follows some basic rules:-
+
+ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
+
+ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
+
+This assumes that the codec PCM output path from the DAC is via a mixer and then
+a PGA (programmable gain amplifier) before being output to the speakers.
+
+
+Minimising Capture Pops and Clicks
+==================================
+
+Capture artifacts are somewhat easier to get rid as we can delay activating the
+ADC until all the pops have occurred. This follows similar power rules to
+playback in that components are powered in a sequence depending upon stream
+startup or shutdown.
+
+ Startup Order - Input PGA --> Mixers --> ADC
+
+ Shutdown Order - ADC --> Mixers --> Input PGA
+
+
+Zipper Noise
+============
+An unwanted zipper noise can occur within the audio playback or capture stream
+when a volume control is changed near its maximum gain value. The zipper noise
+is heard when the gain increase or decrease changes the mean audio signal
+amplitude too quickly. It can be minimised by enabling the zero cross setting
+for each volume control. The ZC forces the gain change to occur when the signal
+crosses the zero amplitude line.
diff --git a/Documentation/sound/oss/ALS b/Documentation/sound/oss/ALS
new file mode 100644
index 00000000..d01ffbfd
--- /dev/null
+++ b/Documentation/sound/oss/ALS
@@ -0,0 +1,66 @@
+ALS-007/ALS-100/ALS-200 based sound cards
+=========================================
+
+Support for sound cards based around the Avance Logic
+ALS-007/ALS-100/ALS-200 chip is included.  These chips are a single
+chip PnP sound solution which is mostly hardware compatible with the
+Sound Blaster 16 card, with most differences occurring in the use of
+the mixer registers.  For this reason the ALS code is integrated
+as part of the Sound Blaster 16 driver (adding only 800 bytes to the
+SB16 driver).
+
+To use an ALS sound card under Linux, enable the following options as
+modules in the sound configuration section of the kernel config:
+  - 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
+  - FM synthesizer (YM3812/OPL-3) support 
+  - standalone MPU401 support may be required for some cards; for the
+    ALS-007, when using isapnptools, it is required
+Since the ALS-007/100/200 are PnP cards, ISAPnP support should probably be
+compiled in.  If kernel level PnP support is not included, isapnptools will
+be required to configure the card before the sound modules are loaded.
+
+When using kernel level ISAPnP, the kernel should correctly identify and
+configure all resources required by the card when the "sb" module is
+inserted.  Note that the ALS-007 does not have a 16 bit DMA channel and that
+the MPU401 interface on this card uses a different interrupt to the audio
+section.  This should all be correctly configured by the kernel; if problems
+with the MPU401 interface surface, try using the standalone MPU401 module,
+passing "0" as the "sb" module's "mpu_io" module parameter to prevent the
+soundblaster driver attempting to register the MPU401 itself.  The onboard
+synth device can be accessed using the "opl3" module.
+
+If isapnptools is used to wake up the sound card (as in 2.2.x), the settings
+of the card's resources should be passed to the kernel modules ("sb", "opl3"
+and "mpu401") using the module parameters.  When configuring an ALS-007, be
+sure to specify different IRQs for the audio and MPU401 sections - this card
+requires they be different.  For "sb", "io", "irq" and "dma" should be set
+to the same values used to configure the audio section of the card with
+isapnp.  "dma16" should be explicitly set to "-1" for an ALS-007 since this
+card does not have a 16 bit dma channel; if not specified the kernel will
+default to using channel 5 anyway which will cause audio not to work. 
+"mpu_io" should be set to 0.  The "io" parameter of the "opl3" module should
+also agree with the setting used by isapnp.  To get the MPU401 interface
+working on an ALS-007 card, the "mpu401" module will be required since this
+card uses separate IRQs for the audio and MPU401 sections and there is no
+parameter available to pass a different IRQ to the "sb" driver (whose
+inbuilt MPU401 driver would otherwise be fine).  Insert the mpu401 module
+passing appropriate values using the "io" and "irq" parameters.
+
+The resulting sound driver will provide the following capabilities:
+  - 8 and 16 bit audio playback
+  - 8 and 16 bit audio recording
+  - Software selection of record source (line in, CD, FM, mic, master)
+  - Record and playback of midi data via the external MPU-401
+  - Playback of midi data using inbuilt FM synthesizer
+  - Control of the ALS-007 mixer via any OSS-compatible mixer programs. 
+    Controls available are Master (L&R), Line in (L&R), CD (L&R), 
+    DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono).
+
+Jonathan Woithe
+jwoithe@physics.adelaide.edu.au
+30 March 1998
+
+Modified 2000-02-26 by Dave Forrest, drf5n@virginia.edu to add ALS100/ALS200
+Modified 2000-04-10 by Paul Laufer, pelaufer@csupomona.edu to add ISAPnP info.
+Modified 2000-11-19 by Jonathan Woithe, jwoithe@physics.adelaide.edu.au
+ - updated information for kernel 2.4.x.
diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16
new file mode 100644
index 00000000..e0dc0641
--- /dev/null
+++ b/Documentation/sound/oss/AudioExcelDSP16
@@ -0,0 +1,101 @@
+Driver
+------
+
+Information about Audio Excel DSP 16 driver can be found in the source
+file aedsp16.c
+Please, read the head of the source before using it. It contain useful
+information.
+
+Configuration
+-------------
+
+The Audio Excel configuration, is now done with the standard Linux setup.
+You have to configure the sound card (Sound Blaster or Microsoft Sound System)
+and, if you want it, the Roland MPU-401 (do not use the Sound Blaster MPU-401,
+SB-MPU401) in the main driver menu. Activate the lowlevel drivers then select
+the Audio Excel hardware that you want to initialize. Check the IRQ/DMA/MIRQ
+of the Audio Excel initialization: it must be the same as the SBPRO (or MSS)
+setup. If the parameters are different, correct it.
+I you own a Gallant's audio card based on SC-6600, activate the SC-6600 support.
+If you want to change the configuration of the sound board, be sure to
+check off all the configuration items before re-configure it.
+
+Module parameters
+-----------------
+To use this driver as a module, you must configure some module parameters, to
+set up I/O addresses, IRQ lines and DMA channels. Some parameters are
+mandatory while some others are optional. Here a list of parameters you can
+use with this module:
+
+Name		Description
+====		===========
+MANDATORY
+io		I/O base address (0x220 or 0x240)
+irq		irq line (5, 7, 9, 10 or 11)
+dma		dma channel (0, 1 or 3)
+
+OPTIONAL
+mss_base	I/O base address for activate MSS mode (default SBPRO)
+		(0x530 or 0xE80)
+mpu_base	I/O base address for activate MPU-401 mode
+		(0x300, 0x310, 0x320 or 0x330)
+mpu_irq		MPU-401 irq line (5, 7, 9, 10 or 0)
+
+The /etc/modprobe.conf will have lines like this:
+
+options opl3 io=0x388
+options ad1848 io=0x530 irq=11 dma=3
+options aedsp16 io=0x220 irq=11 dma=3 mss_base=0x530
+
+Where the aedsp16 options are the options for this driver while opl3 and
+ad1848 are the corresponding options for the MSS and OPL3 modules.
+
+Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly
+the sound card. Installation dependencies must be written in the modprobe.conf
+file:
+
+install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848
+install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3
+
+Then you must load the sound modules stack in this order:
+sound -> aedsp16 -> [ ad1848, opl3 ]
+
+With the above configuration, loading ad1848 or opl3 modules, will
+automatically load all the sound stack.
+
+Sound cards supported
+---------------------
+This driver supports the SC-6000 and SC-6600 based Gallant's sound card.
+It don't support the Audio Excel DSP 16 III (try the SC-6600 code).
+I'm working on the III version of the card: if someone have useful
+information about it, please let me know.
+For all the non-supported audio cards, you have to boot MS-DOS (or WIN95)
+activating the audio card with the MS-DOS device driver, then you have to
+-- and boot Linux.
+Follow these steps:
+
+1) Compile Linux kernel with standard sound driver, using the emulation
+   you want, with the parameters of your audio card,
+   e.g. Microsoft Sound System irq10 dma3
+2) Install your new kernel as the default boot kernel.
+3) Boot MS-DOS and configure the audio card with the boot time device
+   driver, for MSS irq10 dma3 in our example.
+4) -- and boot Linux. This will maintain the DOS configuration
+   and will boot the new kernel with sound driver. The sound driver will find
+   the audio card and will recognize and attach it.
+
+Reports on User successes
+-------------------------
+
+> Date: Mon, 29 Jul 1996 08:35:40 +0100
+> From: Mr S J Greenaway 
+> To: riccardo@cdc8g5.cdc.polimi.it (Riccardo Facchetti)
+> Subject: Re: Audio Excel DSP 16 initialization code
+>
+> Just to let you know got my Audio Excel (emulating a MSS) working
+> with my original SB16, thanks for the driver!
+
+
+Last revised: 20 August 1998
+Riccardo Facchetti
+fizban@tin.it
diff --git a/Documentation/sound/oss/CMI8330 b/Documentation/sound/oss/CMI8330
new file mode 100644
index 00000000..9c439f1a
--- /dev/null
+++ b/Documentation/sound/oss/CMI8330
@@ -0,0 +1,153 @@
+Documentation for CMI 8330 (SoundPRO) 
+-------------------------------------
+Alessandro Zummo 
+
+( Be sure to read Documentation/sound/oss/SoundPro too )
+
+
+This adapter is now directly supported by the sb driver.
+
+ The only thing you have to do is to compile the kernel sound
+support as a module and to enable kernel ISAPnP support,
+as shown below.
+
+
+CONFIG_SOUND=m
+CONFIG_SOUND_SB=m
+
+CONFIG_PNP=y
+CONFIG_ISAPNP=y
+
+
+and optionally:
+
+
+CONFIG_SOUND_MPU401=m
+
+ for MPU401 support.
+
+
+(I suggest you to use "make menuconfig" or "make xconfig" 
+ for a more comfortable configuration editing)
+
+
+
+Then you can do
+
+ modprobe sb
+
+and everything will be (hopefully) configured.
+
+You should get something similar in syslog:
+
+sb: CMI8330 detected.
+sb: CMI8330 sb base located at 0x220
+sb: CMI8330 mpu base located at 0x330
+sb: CMI8330 mail reports to Alessandro Zummo 
+sb: ISAPnP reports CMI 8330 SoundPRO at i/o 0x220, irq 7, dma 1,5
+
+
+
+
+The old documentation file follows for reference
+purposes.
+
+
+How to enable CMI 8330 (SOUNDPRO) soundchip on Linux
+------------------------------------------
+Stefan Laudat 
+
+[Note: The CMI 8338 is unrelated and is supported by cmpci.o]
+
+	
+	In order to use CMI8330 under Linux  you just have to use a proper isapnp.conf, a good isapnp and a little bit of patience.  I use isapnp 1.17, but
+you may get a better one I guess at http://www.roestock.demon.co.uk/isapnptools/.
+
+	Of course you will have to compile kernel sound support as module, as shown below:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+CONFIG_SOUND_ADLIB=m
+CONFIG_SOUND_MPU401=m
+# Mikro$chaft sound system (kinda useful here ;))	
+CONFIG_SOUND_MSS=m
+
+	The /etc/isapnp.conf file will be:
+
+
+
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+(VERIFYLD N)
+
+
+# WSS 
+
+(CONFIGURE CMI0001/16777472 (LD 0
+(IO 0 (SIZE 8) (BASE 0x0530))
+(IO 1 (SIZE 8) (BASE 0x0388))
+(INT 0 (IRQ 7 (MODE +E)))
+(DMA 0 (CHANNEL 0))
+(NAME "CMI0001/16777472[0]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# MPU
+
+(CONFIGURE CMI0001/16777472 (LD 1
+(IO 0 (SIZE 2) (BASE 0x0330))
+(INT 0 (IRQ 11 (MODE +E)))
+(NAME "CMI0001/16777472[1]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# Joystick
+
+(CONFIGURE CMI0001/16777472 (LD 2
+(IO 0 (SIZE 8) (BASE 0x0200))
+(NAME "CMI0001/16777472[2]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# SoundBlaster 
+ 
+(CONFIGURE CMI0001/16777472 (LD 3
+(IO 0 (SIZE 16) (BASE 0x0220))
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 5))
+(NAME "CMI0001/16777472[3]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+
+(WAITFORKEY)
+
+
+
+	The module sequence is trivial:
+
+/sbin/insmod soundcore
+/sbin/insmod sound
+/sbin/insmod uart401
+# insert this first
+/sbin/insmod ad1848 io=0x530 irq=7 dma=0 soundpro=1
+# The sb module is an alternative to the ad1848 (Microsoft Sound System)
+# Anyhow, this is full duplex and has MIDI
+/sbin/insmod sb io=0x220 dma=1 dma16=5 irq=5 mpu_io=0x330
+
+
+
+Alma Chao  suggests the following /etc/modprobe.conf:
+
+alias sound ad1848
+alias synth0 opl3
+options ad1848 io=0x530 irq=7 dma=0 soundpro=1
+options opl3 io=0x388
+
+	
diff --git a/Documentation/sound/oss/ESS b/Documentation/sound/oss/ESS
new file mode 100644
index 00000000..bba93b4d
--- /dev/null
+++ b/Documentation/sound/oss/ESS
@@ -0,0 +1,34 @@
+Documentation for the ESS AudioDrive chips
+
+In 2.4 kernels the SoundBlaster driver not only tries to detect an ESS chip, it
+tries to detect the type of ESS chip too. The correct detection of the chip 
+doesn't always succeed however, so unless you use the kernel isapnp facilities
+(and you chip is pnp capable) the default behaviour is 2.0 behaviour which
+means: only detect ES688 and ES1688.
+
+All ESS chips now have a recording level setting. This is a need-to-have for
+people who want to use their ESS for recording sound.
+
+Every chip that's detected as a later-than-es1688 chip has a 6 bits logarithmic
+master volume control.
+
+Every chip that's detected as a ES1887 now has Full Duplex support. Made a 
+little testprogram that shows that is works, haven't seen a real program that
+needs this however.
+
+For ESS chips an additional parameter "esstype" can be specified. This controls
+the (auto) detection of the ESS chips. It can have 3 kinds of values:
+
+-1   Act like 2.0 kernels: only detect ES688 or ES1688.
+0	 Try to auto-detect the chip (may fail for ES1688)
+688  The chip will be treated as  ES688
+1688  ,,  ,,   ,,  ,,    ,,   ,, ES1688
+1868  ,,  ,,   ,,  ,,    ,,   ,, ES1868
+1869  ,,  ,,   ,,  ,,    ,,   ,, ES1869
+1788  ,,  ,,   ,,  ,,    ,,   ,, ES1788
+1887  ,,  ,,   ,,  ,,    ,,   ,, ES1887
+1888  ,,  ,,   ,,  ,,    ,,   ,, ES1888
+
+Because Full Duplex is supported for ES1887 you can specify a second DMA
+channel by specifying module parameter dma16. It can be one of: 0, 1, 3 or 5.
+
diff --git a/Documentation/sound/oss/ESS1868 b/Documentation/sound/oss/ESS1868
new file mode 100644
index 00000000..55e922f2
--- /dev/null
+++ b/Documentation/sound/oss/ESS1868
@@ -0,0 +1,55 @@
+Documentation for the ESS1868F AudioDrive PnP sound card
+
+The ESS1868 sound card is a PnP ESS1688-compatible 16-bit sound card.
+
+It should be automatically detected by the Linux Kernel isapnp support when you
+load the sb.o module. Otherwise you should take care of:
+
+  *  The ESS1868 does not allow use of a 16-bit DMA, thus DMA 0, 1, 2, and 3
+     may only be used.
+
+  *  isapnptools version 1.14 does work with ESS1868.  Earlier versions might
+     not.
+
+  *  Sound support MUST be compiled as MODULES, not statically linked
+     into the kernel.
+
+
+NOTE: this is only needed when not using the kernel isapnp support!
+
+For configuring the sound card's I/O addresses, IRQ and DMA, here is a
+sample copy of the isapnp.conf directives regarding the ESS1868:
+
+(CONFIGURE ESS1868/-1 (LD 1
+(IO 0 (BASE 0x0220))
+(IO 1 (BASE 0x0388))
+(IO 2 (BASE 0x0330))
+(DMA 0 (CHANNEL 1))
+(INT 0 (IRQ 5 (MODE +E)))
+(ACT Y)
+))
+
+(for a full working isapnp.conf file, remember the
+(ISOLATE)
+(IDENTIFY *)
+at the beginning and the
+(WAITFORKEY)
+at the end.)
+
+In this setup, the main card I/O is 0x0220, FM synthesizer is 0x0388, and
+the MPU-401 MIDI port is located at 0x0330.  IRQ is IRQ 5, DMA is channel 1.
+
+After configuring the sound card via isapnp, to use the card you must load
+the sound modules with the proper I/O information.  Here is my setup:
+
+# ESS1868F AudioDrive initialization
+
+/sbin/modprobe sound
+/sbin/insmod uart401
+/sbin/insmod sb io=0x220 irq=5 dma=1 dma16=-1
+/sbin/insmod mpu401 io=0x330
+/sbin/insmod opl3 io=0x388
+/sbin/insmod v_midi
+
+opl3 is the FM synthesizer
+/sbin/insmod opl3 io=0x388
diff --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction
new file mode 100644
index 00000000..75d967ff
--- /dev/null
+++ b/Documentation/sound/oss/Introduction
@@ -0,0 +1,459 @@
+Introduction	Notes on Modular Sound Drivers and Soundcore
+Wade Hampton 
+2/14/2001
+
+Purpose:  
+========
+This document provides some general notes on the modular 
+sound drivers and their configuration, along with the 
+support modules sound.o and soundcore.o.
+
+Note, some of this probably should be added to the Sound-HOWTO!
+
+Note, soundlow.o was present with 2.2 kernels but is not 
+required for 2.4.x kernels.  References have been removed
+to this.
+
+
+Copying:
+========
+none
+
+
+History:
+========
+0.1.0  11/20/1998  First version, draft
+1.0.0  11/1998     Alan Cox changes, incorporation in 2.2.0
+                   as Documentation/sound/oss/Introduction
+1.1.0  6/30/1999   Second version, added notes on making the drivers,
+                   added info on multiple sound cards of similar types,]
+                   added more diagnostics info, added info about esd.
+                   added info on OSS and ALSA.
+1.1.1  19991031	   Added notes on sound-slot- and sound-service.
+			(Alan Cox)
+1.1.2  20000920    Modified for Kernel 2.4 (Christoph Hellwig)
+1.1.3  20010214    Minor notes and corrections (Wade Hampton)
+                   Added examples of sound-slot-0, etc.
+
+
+Modular Sound Drivers:
+======================
+
+Thanks to the GREAT work by Alan Cox (alan@lxorguk.ukuu.org.uk),
+
+[And Oleg Drokin, Thomas Sailer, Andrew Veliath and more than a few 
+ others - not to mention Hannu's original code being designed well
+ enough to cope with that kind of chopping up](Alan)
+
+the standard Linux kernels support a modular sound driver.  From
+Alan's comments in linux/drivers/sound/README.FIRST:
+
+  The modular sound driver patches were funded by Red Hat Software 
+  (www.redhat.com). The sound driver here is thus a modified version of 
+  Hannu's code. Please bear that in mind when considering the appropriate
+  forums for bug reporting.
+
+The modular sound drivers may be loaded via insmod or modprobe.  
+To support all the various sound modules, there are two general 
+support modules that must be loaded first:
+ 
+   soundcore.o:   Top level handler for the sound system, provides
+                  a set of functions for registration of devices
+                  by type.
+
+   sound.o:       Common sound functions required by all modules.
+
+For the specific sound modules (e.g., sb.o for the Soundblaster), 
+read the documentation on that module to determine what options
+are available, for example IRQ, address, DMA.
+
+Warning, the options for different cards sometime use different names 
+for the same or a similar feature (dma1= versus dma16=).  As a last 
+resort, inspect the code (search for module_param).
+
+Notes:
+
+1.  There is a new OpenSource sound driver called ALSA which is
+    currently under development:  http://www.alsa-project.org/
+    The ALSA drivers support some newer hardware that may not 
+    be supported by this sound driver and also provide some 
+    additional features.
+
+2.  The commercial OSS driver may be obtained from the site:
+    http://www.opensound.com.  This may be used for cards that
+    are unsupported by the kernel driver, or may be used
+    by other operating systems.  
+
+3.  The enlightenment sound daemon may be used for playing
+    multiple sounds at the same time via a single card, eliminating
+    some of the requirements for multiple sound card systems.  For
+    more information, see:  http://www.tux.org/~ricdude/EsounD.html  
+    The "esd" program may be used with the real-player and mpeg 
+    players like mpg123 and x11amp.  The newer real-player 
+    and some games even include built-in support for ESD!
+
+
+Building the Modules:
+=====================
+
+This document does not provide full details on building the 
+kernel, etc.  The notes below apply only to making the kernel
+sound modules.   If this conflicts with the kernel's README,
+the README takes precedence. 
+
+1.  To make the kernel sound modules, cd to your /usr/src/linux
+    directory (typically) and type make config, make menuconfig, 
+    or make xconfig (to start the command line, dialog, or x-based
+    configuration tool).  
+
+2.  Select the Sound option and a dialog will be displayed.  
+
+3.  Select M (module) for "Sound card support".
+
+4.  Select your sound driver(s) as a module.  For ProAudio, Sound
+    Blaster, etc., select M (module) for OSS sound modules.
+    [thanks to Marvin Stodolsky ]A
+
+5.  Make the kernel (e.g., make bzImage), and install the kernel.
+
+6.  Make the modules and install them (make modules; make modules_install).
+
+Note, for 2.5.x kernels, make sure you have the newer module-init-tools 
+installed or modules will not be loaded properly.  2.5.x requires an
+updated module-init-tools.
+
+
+Plug and Play (PnP:
+===================
+
+If the sound card is an ISA PnP card, isapnp may be used
+to configure the card.  See the file isapnp.txt in the 
+directory one level up (e.g., /usr/src/linux/Documentation).
+
+Also the 2.4.x kernels provide PnP capabilities, see the 
+file NEWS in this directory.
+
+PCI sound cards are highly recommended, as they are far 
+easier to configure and from what I have read, they use
+less resources and are more CPU efficient.
+
+
+INSMOD:
+=======
+
+If loading via insmod, the common modules must be loaded in the 
+order below BEFORE loading the other sound modules.  The card-specific
+modules may then be loaded (most require parameters).  For example,
+I use the following via a shell script to load my SoundBlaster:
+
+SB_BASE=0x240
+SB_IRQ=9
+SB_DMA=3
+SB_DMA2=5
+SB_MPU=0x300
+#
+echo Starting sound
+/sbin/insmod soundcore
+/sbin/insmod sound  
+#
+echo Starting sound blaster....
+/sbin/insmod uart401
+/sbin/insmod sb io=$SB_BASE irq=$SB_IRQ dma=$SB_DMA dma16=$SB_DMA2 mpu_io=$SB_MP
+
+When using sound as a module, I typically put these commands
+in a file such as /root/soundon.sh.
+
+
+MODPROBE:
+=========
+
+If loading via modprobe, these common files are automatically loaded 
+when requested by modprobe.  For example, my /etc/modprobe.conf contains:
+
+alias sound sb 
+options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300
+
+All you need to do to load the module is:
+
+	/sbin/modprobe sb
+
+
+Sound Status:
+=============
+
+The status of sound may be read/checked by:
+        cat (anyfile).au >/dev/audio
+
+[WWH:  This may not work properly for SoundBlaster PCI 128 cards
+such as the es1370/1 (see the es1370/1 files in this directory) 
+as they do not automatically support uLaw on /dev/audio.]
+
+The status of the modules and which modules depend on 
+which other modules may be checked by:
+	/sbin/lsmod
+
+/sbin/lsmod should show something like the following:
+	sb                     26280   0 
+	uart401                 5640   0  [sb]
+	sound                  57112   0  [sb uart401]
+	soundcore               1968   8  [sb sound]
+
+
+Removing Sound: 
+=============== 
+
+Sound may be removed by using /sbin/rmmod in the reverse order
+in which you load the modules.  Note, if a program has a sound device
+open (e.g., xmixer), that module (and the modules on which it 
+depends) may not be unloaded.
+
+For example, I use the following to remove my Soundblaster (rmmod
+in the reverse order in which I loaded the modules):
+
+/sbin/rmmod sb
+/sbin/rmmod uart401
+/sbin/rmmod sound
+/sbin/rmmod soundcore
+
+When using sound as a module, I typically put these commands
+in a script such as /root/soundoff.sh.
+
+
+Removing Sound for use with OSS: 
+================================ 
+
+If you get really stuck or have a card that the kernel modules
+will not support, you can get a commercial sound driver from
+http://www.opensound.com.  Before loading the commercial sound
+driver, you should do the following:
+
+1.  remove sound modules (detailed above)
+2.  remove the sound modules from /etc/modprobe.conf
+3.  move the sound modules from /lib/modules//misc
+    (for example, I make a /lib/modules//misc/tmp
+    directory and copy the sound module files to that 
+    directory).
+
+
+Multiple Sound Cards:
+=====================
+
+The sound drivers will support multiple sound cards and there
+are some great applications like multitrack that support them.  
+Typically, you need two sound cards of different types.  Note, this
+uses more precious interrupts and DMA channels and sometimes 
+can be a configuration nightmare.  I have heard reports of 3-4
+sound cards (typically I only use 2).  You can sometimes use
+multiple PCI sound cards of the same type.
+
+On my machine I have two sound cards (cs4232 and Soundblaster Vibra
+16).  By loading sound as modules, I can control which is the first
+sound device (/dev/dsp, /dev/audio, /dev/mixer) and which is 
+the second.  Normally, the cs4232 (Dell sound on the motherboard) 
+would be the first sound device, but I prefer the Soundblaster.  
+All you have to do is to load the one you want as /dev/dsp 
+first (in my case "sb") and then load the other one
+(in my case "cs4232").
+
+If you have two cards of the same type that are jumpered 
+cards or different PnP revisions, you may load the same 
+module twice.  For example, I have a SoundBlaster vibra 16
+and an older SoundBlaster 16 (jumpers).  To load the module
+twice, you need to do the following:
+
+1.  Copy the sound modules to a new name.  For example
+    sb.o could be copied (or symlinked) to sb1.o for the
+    second SoundBlaster.
+
+2.  Make a second entry in /etc/modprobe.conf, for example,
+    sound1 or sb1.  This second entry should refer to the
+    new module names for example sb1, and should include
+    the I/O, etc. for the second sound card.
+
+3.  Update your soundon.sh script, etc.
+
+Warning:  I have never been able to get two PnP sound cards of the
+same type to load at the same time.  I have tried this several times
+with the Soundblaster Vibra 16 cards.  OSS has indicated that this
+is a PnP problem....  If anyone has any luck doing this, please 
+send me an E-MAIL.  PCI sound cards should not have this problem.a
+Since this was originally release, I have received a couple of 
+mails from people who have accomplished this!
+
+NOTE: In Linux 2.4 the Sound Blaster driver (and only this one yet)
+supports multiple cards with one module by default.
+Read the file 'Soundblaster' in this directory for details.
+
+
+Sound Problems:
+===============
+
+First RTFM (including the troubleshooting section 
+in the Sound-HOWTO). 
+
+1)  If you are having problems loading the modules (for
+    example, if you get device conflict errors) try the
+    following:
+
+  A)  If you have Win95 or NT on the same computer,  
+      write down what addresses, IRQ, and DMA channels
+      those were using for the same hardware.  You probably 
+      can use these addresses, IRQs, and DMA channels.
+      You should really do this BEFORE attempting to get
+      sound working!
+  
+  B)  Check (cat) /proc/interrupts, /proc/ioports,
+      and /proc/dma.  Are you trying to use an address,
+      IRQ or DMA port that another device is using?
+  
+  C)  Check (cat) /proc/isapnp
+  
+  D)  Inspect your /var/log/messages file.  Often that will 
+      indicate what IRQ or IO port could not be obtained.
+  
+  E)  Try another port or IRQ.  Note this may involve 
+      using the PnP tools to move the sound card to 
+      another location.  Sometimes this is the only way 
+      and it is more or less trial and error.
+
+2)  If you get motor-boating (the same sound or part of a 
+    sound clip repeated), you probably have either an IRQ
+    or DMA conflict.  Move the card to another IRQ or DMA
+    port.  This has happened to me when playing long files 
+    when I had an IRQ conflict.
+
+3.  If you get dropouts or pauses when playing high sample
+    rate files such as using mpg123 or x11amp/xmms, you may 
+    have too slow of a CPU and may have to use the options to 
+    play the files at 1/2 speed.  For example, you may use
+    the -2 or -4 option on mpg123.  You may also get this
+    when trying to play mpeg files stored on a CD-ROM
+    (my Toshiba T8000 PII/366 sometimes has this problem).
+
+4.  If you get "cannot access device" errors, your /dev/dsp
+    files, etc. may be set to owner root, mode 600.  You 
+    may have to use the command:
+      chmod 666 /dev/dsp /dev/mixer /dev/audio
+
+5.  If you get "device busy" errors, another program has the
+    sound device open.  For example, if using the Enlightenment
+    sound daemon "esd", the "esd" program has the sound device.
+    If using "esd", please RTFM the docs on ESD.  For example,
+    esddsp  may be used to play files via a non-esd
+    aware program.
+
+6)  Ask for help on the sound list or send E-MAIL to the
+    sound driver author/maintainer.
+
+7)  Turn on debug in drivers/sound/sound_config.h (DEB, DDB, MDB).
+
+8)  If the system reports insufficient DMA memory then you may want to
+    load sound with the "dmabufs=1" option. Or in /etc/conf.modules add
+	
+	preinstall sound dmabufs=1
+
+    This makes the sound system allocate its buffers and hang onto them.
+
+    You may also set persistent DMA when building a 2.4.x kernel.
+
+
+Configuring Sound:
+==================
+
+There are several ways of configuring your sound:
+
+1)  On the kernel command line (when using the sound driver(s)
+    compiled in the kernel). Check the driver source and
+    documentation for details.
+
+2)  On the command line when using insmod or in a bash script
+    using command line calls to load sound.
+
+3)  In /etc/modprobe.conf when using modprobe.
+
+4)  Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based).
+
+5)  Via the OSS soundconf program (with the commercial version
+    of the OSS driver.
+
+6)  By just loading the module and let isapnp do everything relevant
+    for you. This works only with a few drivers yet and - of course -
+    only with isapnp hardware.
+
+And I am sure, several other ways.  
+
+Anyone want to write a linuxconf module for configuring sound?
+
+
+Module Loading:
+===============
+
+When a sound card is first referenced and sound is modular, the sound system
+will ask for the sound devices to be loaded. Initially it requests that
+the driver for the sound system is loaded. It then will ask for 
+sound-slot-0, where 0 is the first sound card. (sound-slot-1 the second and
+so on). Thus you can do
+
+alias sound-slot-0 sb
+
+To load a soundblaster at this point. If the slot loading does not provide
+the desired device - for example a soundblaster does not directly provide
+a midi synth in all cases then it will request "sound-service-0-n" where n
+is
+
+  0	Mixer
+
+  2  	MIDI
+
+  3, 4	DSP audio
+
+
+For example, I use the following to load my Soundblaster PCI 128
+(ES 1371) card first, followed by my SoundBlaster Vibra 16 card,
+then by my TV card:
+
+# Load the Soundblaster PCI 128 as /dev/dsp, /dev/dsp1, /dev/mixer
+alias sound-slot-0 es1371
+
+# Load the Soundblaster Vibra 16 as /dev/dsp2, /dev/mixer1
+alias sound-slot-1 sb
+options sb io=0x240 irq=5 dma=1 dma16=5 mpu_io=0x330
+
+# Load the BTTV (TV card) as /dev/mixer2
+alias sound-slot-2 bttv
+alias sound-service-2-0 tvmixer
+
+pre-install bttv  modprobe tuner ; modprobe tvmixer
+pre-install tvmixer modprobe msp3400; modprobe tvaudio 
+options tuner debug=0 type=8 
+options bttv  card=0 radio=0 pll=0
+
+
+For More Information (RTFM):
+============================
+1)  Information on kernel modules: manual pages for insmod and modprobe.
+
+2)  Information on PnP, RTFM manual pages for isapnp.
+
+3)  Sound-HOWTO and Sound-Playing-HOWTO.
+
+4)  OSS's WWW site at http://www.opensound.com.
+
+5)  All the files in Documentation/sound.
+
+6)  The comments and code in linux/drivers/sound.
+
+7)  The sndconfig and rhsound documentation from Red Hat.
+
+8)  The Linux-sound mailing list:  sound-list@redhat.com.
+
+9)  Enlightenment documentation (for info on esd)
+    http://www.tux.org/~ricdude/EsounD.html.
+
+10) ALSA home page:  http://www.alsa-project.org/
+
+
+Contact Information:
+====================
+Wade Hampton:  (whampton@staffnet.com)
+
diff --git a/Documentation/sound/oss/MultiSound b/Documentation/sound/oss/MultiSound
new file mode 100644
index 00000000..e4a18bb7
--- /dev/null
+++ b/Documentation/sound/oss/MultiSound
@@ -0,0 +1,1137 @@
+#! /bin/sh
+#
+#  Turtle Beach MultiSound Driver Notes
+#  -- Andrew Veliath 
+#
+#  Last update:                      September 10, 1998
+#  Corresponding msnd driver:        0.8.3
+#
+# ** This file is a README (top part) and shell archive (bottom part).
+#    The corresponding archived utility sources can be unpacked by
+#    running `sh MultiSound' (the utilities are only needed for the
+#    Pinnacle and Fiji cards). **
+#
+#
+#  -=-=- Getting Firmware -=-=-
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  See the section `Obtaining and Creating Firmware Files' in this
+#  document for instructions on obtaining the necessary firmware
+#  files.
+#  
+#  
+#  Supported Features
+#  ~~~~~~~~~~~~~~~~~~
+#  
+#  Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is
+#  not currently available) and mixer functionality (/dev/mixer) are
+#  supported (memory mapped digital audio is not yet supported).
+#  Digital transfers and monitoring can be done as well if you have
+#  the digital daughterboard (see the section on using the S/PDIF port
+#  for more information).
+#
+#  Support for the Turtle Beach MultiSound Hurricane architecture is
+#  composed of the following modules (these can also operate compiled
+#  into the kernel):
+#  
+#  msnd               - MultiSound base (requires soundcore)
+#
+#  msnd_classic       - Base audio/mixer support for Classic, Monetery and
+#                       Tahiti cards
+#
+#  msnd_pinnacle      - Base audio/mixer support for Pinnacle and Fiji cards
+#  
+#  
+#  Important Notes - Read Before Using
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  The firmware files are not included (may change in future).  You
+#  must obtain these images from Turtle Beach (they are included in
+#  the MultiSound Development Kits), and place them in /etc/sound for
+#  example, and give the full paths in the Linux configuration.  If
+#  you are compiling in support for the MultiSound driver rather than
+#  using it as a module, these firmware files must be accessible
+#  during kernel compilation.
+#
+#  Please note these files must be binary files, not assembler.  See
+#  the section later in this document for instructions to obtain these
+#  files.
+#  
+#  
+#  Configuring Card Resources
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  ** This section is very important, as your card may not work at all
+#     or your machine may crash if you do not do this correctly. **
+#
+#  * Classic/Monterey/Tahiti
+#  
+#  These cards are configured through the driver msnd_classic.  You must
+#  know the io port, then the driver will select the irq and memory resources
+#  on the card.  It is up to you to know if these are free locations or now,
+#  a conflict can lock the machine up.
+#
+#  * Pinnacle/Fiji
+#
+#  The Pinnacle and Fiji cards have an extra config port, either
+#  0x250, 0x260 or 0x270.  This port can be disabled to have the card
+#  configured strictly through PnP, however you lose the ability to
+#  access the IDE controller and joystick devices on this card when
+#  using PnP.  The included pinnaclecfg program in this shell archive
+#  can be used to configure the card in non-PnP mode, and in PnP mode
+#  you can use isapnptools.  These are described briefly here.
+#
+#  pinnaclecfg is not required; you can use the msnd_pinnacle module
+#  to fully configure the card as well.  However, pinnaclecfg can be
+#  used to change the resource values of a particular device after the
+#  msnd_pinnacle module has been loaded.  If you are compiling the
+#  driver into the kernel, you must set these values during compile
+#  time, however other peripheral resource values can be changed with
+#  the pinnaclecfg program after the kernel is loaded.
+#
+#
+#  *** PnP mode
+#  
+#  Use pnpdump to obtain a sample configuration if you can; I was able
+#  to obtain one with the command `pnpdump 1 0x203' -- this may vary
+#  for you (running pnpdump by itself did not work for me).  Then,
+#  edit this file and use isapnp to uncomment and set the card values.
+#  Use these values when inserting the msnd_pinnacle module.  Using
+#  this method, you can set the resources for the DSP and the Kurzweil
+#  synth (Pinnacle).  Since Linux does not directly support PnP
+#  devices, you may have difficulty when using the card in PnP mode
+#  when it the driver is compiled into the kernel.  Using non-PnP mode
+#  is preferable in this case.
+#
+#  Here is an example mypinnacle.conf for isapnp that sets the card to
+#  io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil
+#  synth to 0x330 and irq 9 (may need editing for your system):
+#
+#  (READPORT 0x0203)
+#  (CSN 2)
+#  (IDENTIFY *)
+#  
+#  # DSP
+#  (CONFIGURE BVJ0440/-1 (LD 0
+#          (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000))
+#          (ACT Y)))
+#  
+#  # Kurzweil Synth (Pinnacle Only)
+#  (CONFIGURE BVJ0440/-1 (LD 1
+#          (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E)))
+#          (ACT Y)))
+#  
+#  (WAITFORKEY)
+#
+#
+#  *** Non-PnP mode
+#  
+#  The second way is by running the card in non-PnP mode.  This
+#  actually has some advantages in that you can access some other
+#  devices on the card, such as the joystick and IDE controller.  To
+#  configure the card, unpack this shell archive and build the
+#  pinnaclecfg program.  Using this program, you can assign the
+#  resource values to the card's devices, or disable the devices.  As
+#  an alternative to using pinnaclecfg, you can specify many of the
+#  configuration values when loading the msnd_pinnacle module (or
+#  during kernel configuration when compiling the driver into the
+#  kernel).
+#
+#  If you specify cfg=0x250 for the msnd_pinnacle module, it
+#  automatically configure the card to the given io, irq and memory
+#  values using that config port (the config port is jumper selectable
+#  on the card to 0x250, 0x260 or 0x270).
+#
+#  See the `msnd_pinnacle Additional Options' section below for more
+#  information on these parameters (also, if you compile the driver
+#  directly into the kernel, these extra parameters can be useful
+#  here).
+#
+#
+# ** It is very easy to cause problems in your machine if you choose a
+#    resource value which is incorrect. **
+#  
+#
+#  Examples
+#  ~~~~~~~~
+#  
+#  * MultiSound Classic/Monterey/Tahiti:
+#  
+#  modprobe soundcore
+#  insmod msnd
+#  insmod msnd_classic io=0x290 irq=7 mem=0xd0000
+#  
+#  * MultiSound Pinnacle in PnP mode:
+#  
+#  modprobe soundcore
+#  insmod msnd
+#  isapnp mypinnacle.conf
+#  insmod msnd_pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values
+#  
+#  * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port,
+#    one of 0x250, 0x260 or 0x270):
+#  
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000
+#  
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP
+#   mode, add the following (assumes you did `isapnp mypinnacle.conf'):
+#  
+#  insmod sound
+#  insmod mpu401 io=0x330 irq=9                    <-- match mypinnacle.conf values
+#  
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP
+#   mode, add the following.  Note how we first configure the peripheral's
+#   resources, _then_ install a Linux driver for it:
+#  
+#  insmod sound
+#  pinnaclecfg 0x250 mpu 0x330 9
+#  insmod mpu401 io=0x330 irq=9
+#
+#  -- OR you can use the following sequence without pinnaclecfg in non-PnP mode:
+#
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9
+#  insmod sound
+#  insmod mpu401 io=0x330 irq=9
+#
+# * To setup the joystick port on the Pinnacle in non-PnP mode (though
+#   you have to find the actual Linux joystick driver elsewhere), you
+#   can use pinnaclecfg:
+#
+#   pinnaclecfg 0x250 joystick 0x200
+#
+#  -- OR you can configure this using msnd_pinnacle with the following:
+#
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200
+#
+#  
+#  msnd_classic, msnd_pinnacle Required Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  If the following options are not given, the module will not load.
+#  Examine the kernel message log for informative error messages.
+#  WARNING--probing isn't supported so try to make sure you have the
+#  correct shared memory area, otherwise you may experience problems.
+#  
+#  io                   I/O base of DSP, e.g. io=0x210
+#  irq                  IRQ number, e.g. irq=5
+#  mem                  Shared memory area, e.g. mem=0xd8000
+#  
+#  
+#  msnd_classic, msnd_pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  fifosize             The digital audio FIFOs, in kilobytes.  If not
+#                       specified, the default will be used.  Increasing
+#                       this value will reduce the chance of a FIFO
+#                       underflow at the expense of increasing overall
+#                       latency.  For example, fifosize=512 will
+#                       allocate 512kB read and write FIFOs (1MB total).
+#                       While this may reduce dropouts, a heavy machine
+#                       load will undoubtedly starve the FIFO of data
+#                       and you will eventually get dropouts.  One
+#                       option is to alter the scheduling priority of
+#                       the playback process, using `nice' or some form
+#                       of POSIX soft real-time scheduling.
+#
+#  calibrate_signal     Setting this to one calibrates the ADCs to the
+#                       signal, zero calibrates to the card (defaults
+#                       to zero).
+#  
+#  
+#  msnd_pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  digital              Specify digital=1 to enable the S/PDIF input
+#                       if you have the digital daughterboard
+#                       adapter. This will enable access to the
+#                       DIGITAL1 input for the soundcard in the mixer.
+#                       Some mixer programs might have trouble setting
+#                       the DIGITAL1 source as an input.  If you have
+#                       trouble, you can try the setdigital.c program
+#                       at the bottom of this document.
+#
+#  cfg                  Non-PnP configuration port for the Pinnacle
+#                       and Fiji (typically 0x250, 0x260 or 0x270,
+#                       depending on the jumper configuration).  If
+#                       this option is omitted, then it is assumed
+#                       that the card is in PnP mode, and that the
+#                       specified DSP resource values are already
+#                       configured with PnP (i.e. it won't attempt to
+#                       do any sort of configuration).
+#
+#  When the Pinnacle is in non-PnP mode, you can use the following
+#  options to configure particular devices.  If a full specification
+#  for a device is not given, then the device is not configured.  Note
+#  that you still must use a Linux driver for any of these devices
+#  once their resources are setup (such as the Linux joystick driver,
+#  or the MPU401 driver from OSS for the Kurzweil synth).
+#
+#  mpu_io               I/O port of MPU (on-board Kurzweil synth)
+#  mpu_irq              IRQ of MPU (on-board Kurzweil synth)
+#  ide_io0		First I/O port of IDE controller
+#  ide_io1		Second I/O port of IDE controller
+#  ide_irq		IRQ IDE controller
+#  joystick_io          I/O port of joystick
+#  
+#  
+#  Obtaining and Creating Firmware Files
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#       For the Classic/Tahiti/Monterey
+#       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 
+#  
+#  Download to /tmp and unzip the following file from Turtle Beach:
+#  
+#       ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip
+#  
+#  When unzipped, unzip the file named MsndFiles.zip.  Then copy the
+#  following firmware files to /etc/sound (note the file renaming):
+#  
+#    cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin
+#    cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin
+#  
+#  When configuring the Linux kernel, specify /etc/sound/msndinit.bin and
+#  /etc/sound/msndperm.bin for the two firmware files (Linux kernel
+#  versions older than 2.2 do not ask for firmware paths, and are
+#  hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#  
+#  
+#       For the Pinnacle/Fiji
+#       ~~~~~~~~~~~~~~~~~~~~~
+#  
+#  Download to /tmp and unzip the following file from Turtle Beach (be
+#  sure to use the entire URL; some have had trouble navigating to the
+#  URL):
+#  
+#       ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip
+#
+#  Unpack this shell archive, and run make in the created directory
+#  (you need a C compiler and flex to build the utilities).  This
+#  should give you the executables conv, pinnaclecfg and setdigital.
+#  conv is only used temporarily here to create the firmware files,
+#  while pinnaclecfg is used to configure the Pinnacle or Fiji card in
+#  non-PnP mode, and setdigital can be used to set the S/PDIF input on
+#  the mixer (pinnaclecfg and setdigital should be copied to a
+#  convenient place, possibly run during system initialization).
+#
+#  To generating the firmware files with the `conv' program, we create
+#  the binary firmware files by doing the following conversion
+#  (assuming the archive unpacked into a directory named PINNDDK):
+#  
+#    ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin
+#    ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin
+#  
+#  The conv (and conv.l) program is not needed after conversion and can
+#  be safely deleted.  Then, when configuring the Linux kernel, specify
+#  /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two
+#  firmware files (Linux kernel versions older than 2.2 do not ask for
+#  firmware paths, and are hardcoded to /etc/sound).
+#  
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#  
+#  Using Digital I/O with the S/PDIF Port
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If you have a Pinnacle or Fiji with the digital daughterboard and
+#  want to set it as the input source, you can use this program if you
+#  have trouble trying to do it with a mixer program (be sure to
+#  insert the module with the digital=1 option, or say Y to the option
+#  during compiled-in kernel operation).  Upon selection of the S/PDIF
+#  port, you should be able monitor and record from it.
+#
+#  There is something to note about using the S/PDIF port.  Digital
+#  timing is taken from the digital signal, so if a signal is not
+#  connected to the port and it is selected as recording input, you
+#  will find PCM playback to be distorted in playback rate.  Also,
+#  attempting to record at a sampling rate other than the DAT rate may
+#  be problematic (i.e. trying to record at 8000Hz when the DAT signal
+#  is 44100Hz).  If you have a problem with this, set the recording
+#  input to analog if you need to record at a rate other than that of
+#  the DAT rate.
+#
+#
+#  -- Shell archive attached below, just run `sh MultiSound' to extract.
+#     Contains Pinnacle/Fiji utilities to convert firmware, configure
+#     in non-PnP mode, and select the DIGITAL1 input for the mixer.
+#
+#
+#!/bin/sh
+# This is a shell archive (produced by GNU sharutils 4.2).
+# To extract the files from this archive, save it to some FILE, remove
+# everything before the `!/bin/sh' line above, then type `sh FILE'.
+#
+# Made on 1998-12-04 10:07 EST by .
+# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'.
+#
+# Existing files will *not* be overwritten unless `-c' is specified.
+#
+# This shar contains:
+# length mode       name
+# ------ ---------- ------------------------------------------
+#   2046 -rw-rw-r-- MultiSound.d/setdigital.c
+#  10235 -rw-rw-r-- MultiSound.d/pinnaclecfg.c
+#    106 -rw-rw-r-- MultiSound.d/Makefile
+#    141 -rw-rw-r-- MultiSound.d/conv.l
+#   1472 -rw-rw-r-- MultiSound.d/msndreset.c
+#
+save_IFS="${IFS}"
+IFS="${IFS}:"
+gettext_dir=FAILED
+locale_dir=FAILED
+first_param="$1"
+for dir in $PATH
+do
+  if test "$gettext_dir" = FAILED && test -f $dir/gettext \
+     && ($dir/gettext --version >/dev/null 2>&1)
+  then
+    set `$dir/gettext --version 2>&1`
+    if test "$3" = GNU
+    then
+      gettext_dir=$dir
+    fi
+  fi
+  if test "$locale_dir" = FAILED && test -f $dir/shar \
+     && ($dir/shar --print-text-domain-dir >/dev/null 2>&1)
+  then
+    locale_dir=`$dir/shar --print-text-domain-dir`
+  fi
+done
+IFS="$save_IFS"
+if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED
+then
+  echo=echo
+else
+  TEXTDOMAINDIR=$locale_dir
+  export TEXTDOMAINDIR
+  TEXTDOMAIN=sharutils
+  export TEXTDOMAIN
+  echo="$gettext_dir/gettext -s"
+fi
+touch -am 1231235999 $$.touch >/dev/null 2>&1
+if test ! -f 1231235999 && test -f $$.touch; then
+  shar_touch=touch
+else
+  shar_touch=:
+  echo
+  $echo 'WARNING: not restoring timestamps.  Consider getting and'
+  $echo "installing GNU \`touch', distributed in GNU File Utilities..."
+  echo
+fi
+rm -f 1231235999 $$.touch
+#
+if mkdir _sh01426; then
+  $echo 'x -' 'creating lock directory'
+else
+  $echo 'failed to create lock directory'
+  exit 1
+fi
+# ============= MultiSound.d/setdigital.c ==============
+if test ! -d 'MultiSound.d'; then
+  $echo 'x -' 'creating directory' 'MultiSound.d'
+  mkdir 'MultiSound.d'
+fi
+if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' &&
+/*********************************************************************
+X *
+X * setdigital.c - sets the DIGITAL1 input for a mixer
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X	unsigned long recmask, recsrc;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: setdigital \n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) {
+X		fprintf(stderr, "error: ioctl read recording mask failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (!(recmask & SOUND_MASK_DIGITAL1)) {
+X		fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl read recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	recsrc |= SOUND_MASK_DIGITAL1;
+X	
+X	if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl write recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X	
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' &&
+  chmod 0664 'MultiSound.d/setdigital.c' ||
+  $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed'
+e87217fc3e71288102ba41fd81f71ec4  MultiSound.d/setdigital.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`"
+    test 2046 -eq "$shar_count" ||
+    $echo 'MultiSound.d/setdigital.c:' 'original size' '2046,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/pinnaclecfg.c ==============
+if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' &&
+/*********************************************************************
+X *
+X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program
+X *
+X * This is for NON-PnP mode only.  For PnP mode, use isapnptools.
+X *
+X * This is Linux-specific, and must be run with root permissions.
+X *
+X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+X
+#define IREG_LOGDEVICE		0x07
+#define IREG_ACTIVATE		0x30
+#define LD_ACTIVATE		0x01
+#define LD_DISACTIVATE		0x00
+#define IREG_EECONTROL		0x3F
+#define IREG_MEMBASEHI		0x40
+#define IREG_MEMBASELO		0x41
+#define IREG_MEMCONTROL		0x42
+#define IREG_MEMRANGEHI		0x43
+#define IREG_MEMRANGELO		0x44
+#define MEMTYPE_8BIT		0x00
+#define MEMTYPE_16BIT		0x02
+#define MEMTYPE_RANGE		0x00
+#define MEMTYPE_HIADDR		0x01
+#define IREG_IO0_BASEHI		0x60
+#define IREG_IO0_BASELO		0x61
+#define IREG_IO1_BASEHI		0x62
+#define IREG_IO1_BASELO		0x63
+#define IREG_IRQ_NUMBER		0x70
+#define IREG_IRQ_TYPE		0x71
+#define IRQTYPE_HIGH		0x02
+#define IRQTYPE_LOW		0x00
+#define IRQTYPE_LEVEL		0x01
+#define IRQTYPE_EDGE		0x00
+X
+#define HIBYTE(w)		((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w)		((BYTE)(w))
+#define MAKEWORD(low,hi)	((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+X
+typedef __u8			BYTE;
+typedef __u16			USHORT;
+typedef __u16			WORD;
+X
+static int config_port = -1;
+X
+static int msnd_write_cfg(int cfg, int reg, int value)
+{
+X	outb(reg, cfg);
+X	outb(value, cfg + 1);
+X	if (value != inb(cfg + 1)) {
+X		fprintf(stderr, "error: msnd_write_cfg: I/O error\n");
+X		return -EIO;
+X	}
+X	return 0;
+}
+X
+static int msnd_read_cfg(int cfg, int reg)
+{
+X	outb(reg, cfg);
+X	return inb(cfg + 1);
+}
+X
+static int msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io0(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO0_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io1(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO1_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_irq(int cfg, int num, WORD *irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	
+X	*irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER);
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+X	WORD wmem;
+X
+X	mem >>= 8;
+X	mem &= 0xfff;
+X	wmem = (WORD)mem;
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+X		return -EIO;
+X	if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_mem(int cfg, int num, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	
+X	*mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO),
+X			msnd_read_cfg(cfg, IREG_MEMBASEHI));
+X	*mem <<= 8;
+X
+X	return 0;
+}
+X
+static int msnd_activate_logical(int cfg, int num)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_write_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_write_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_write_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	if (msnd_activate_logical(cfg, num))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_read_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_read_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_read_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_read_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	return 0;
+}
+X
+static void usage(void)
+{
+X	fprintf(stderr,
+X		"\n"
+X		"pinnaclecfg 1.0\n"
+X		"\n"
+X		"usage: pinnaclecfg  [device config]\n"
+X		"\n"
+X		"This is for use with the card in NON-PnP mode only.\n"
+X		"\n"
+X		"Available devices (not all available for Fiji):\n"
+X		"\n"
+X		"        Device                       Description\n"
+X		"        -------------------------------------------------------------------\n"
+X		"        reset                        Reset all devices (i.e. disable)\n"
+X		"        show                         Display current device configurations\n"
+X		"\n"
+X		"        dsp            Audio device\n"
+X		"        mpu                 Internal Kurzweil synth\n"
+X		"        ide           On-board IDE controller\n"
+X		"        joystick                 Joystick port\n"
+X		"\n");
+X	exit(1);
+}
+X
+static int cfg_reset(void)
+{
+X	int i;
+X
+X	for (i = 0; i < 4; ++i)
+X		msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0);
+X	
+X	return 0;
+}
+X
+static int cfg_show(void)
+{
+X	int i;
+X	int count = 0;
+X
+X	for (i = 0; i < 4; ++i) {
+X		WORD io0, io1, irq;
+X		int mem;
+X		msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem);
+X		switch (i) {
+X		case 0:
+X			if (io0 || irq || mem) {
+X				printf("dsp 0x%x %d 0x%x\n", io0, irq, mem);
+X				++count;
+X			}
+X			break;
+X		case 1:
+X			if (io0 || irq) {
+X				printf("mpu 0x%x %d\n", io0, irq);
+X				++count;
+X			}
+X			break;
+X		case 2:
+X			if (io0 || io1 || irq) {
+X				printf("ide 0x%x 0x%x %d\n", io0, io1, irq);
+X				++count;
+X			}
+X			break;
+X		case 3:
+X			if (io0) {
+X				printf("joystick 0x%x\n", io0);
+X				++count;
+X			}
+X			break;
+X		}
+X	}
+X
+X	if (count == 0)
+X		fprintf(stderr, "no devices configured\n");
+X	
+X	return 0;
+}
+X
+static int cfg_dsp(int argc, char *argv[])
+{
+X	int io, irq, mem;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1 ||
+X	    sscanf(argv[2], "0x%x", &mem) != 1)
+X		usage();
+X
+X	if (!(io == 0x290 ||
+X	      io == 0x260 ||
+X	      io == 0x250 ||
+X	      io == 0x240 ||
+X	      io == 0x230 ||
+X	      io == 0x220 ||
+X	      io == 0x210 ||
+X	      io == 0x3e0)) {
+X		fprintf(stderr, "error: io must be one of "
+X			"210, 220, 230, 240, 250, 260, 290, or 3E0\n");
+X		usage();
+X	}
+X	
+X	if (!(irq == 5 ||
+X	      irq == 7 ||
+X	      irq == 9 ||
+X	      irq == 10 ||
+X	      irq == 11 ||
+X	      irq == 12)) {
+X		fprintf(stderr, "error: irq must be one of "
+X			"5, 7, 9, 10, 11 or 12\n");
+X		usage();
+X	}
+X
+X	if (!(mem == 0xb0000 ||
+X	      mem == 0xc8000 ||
+X	      mem == 0xd0000 ||
+X	      mem == 0xd8000 ||
+X	      mem == 0xe0000 ||
+X	      mem == 0xe8000)) {
+X		fprintf(stderr, "error: mem must be one of "
+X			"0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+X		usage();
+X	}
+X
+X	return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem);
+}
+X
+static int cfg_mpu(int argc, char *argv[])
+{
+X	int io, irq;
+X
+X	if (argc < 2 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X	
+X	return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0);
+}
+X
+static int cfg_ide(int argc, char *argv[])
+{
+X	int io0, io1, irq;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io0) != 1 ||
+X	    sscanf(argv[0], "0x%x", &io1) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X	
+X	return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0);
+}
+X
+static int cfg_joystick(int argc, char *argv[])
+{
+X	int io;
+X
+X	if (argc < 1 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1)
+X		usage();
+X	
+X	return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0);
+}
+X
+int main(int argc, char *argv[])
+{
+X	char *device;
+X	int rv = 0;
+X
+X	--argc; ++argv;
+X
+X	if (argc < 2)
+X		usage();
+X
+X	sscanf(argv[0], "0x%x", &config_port);
+X	if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) {
+X		fprintf(stderr, "error:  must be 0x250, 0x260 or 0x270\n");
+X		exit(1);
+X	}
+X	if (ioperm(config_port, 2, 1)) {
+X		perror("ioperm");
+X		fprintf(stderr, "note: pinnaclecfg must be run as root\n");
+X		exit(1);
+X	}
+X	device = argv[1];
+X
+X	argc -= 2; argv += 2;
+X
+X	if (strcmp(device, "reset") == 0)
+X		rv = cfg_reset();
+X	else if (strcmp(device, "show") == 0)
+X		rv = cfg_show();
+X	else if (strcmp(device, "dsp") == 0)
+X		rv = cfg_dsp(argc, argv);
+X	else if (strcmp(device, "mpu") == 0)
+X		rv = cfg_mpu(argc, argv);
+X	else if (strcmp(device, "ide") == 0)
+X		rv = cfg_ide(argc, argv);
+X	else if (strcmp(device, "joystick") == 0)
+X		rv = cfg_joystick(argc, argv);
+X	else {
+X		fprintf(stderr, "error: unknown device %s\n", device);
+X		usage();
+X	}
+X
+X	if (rv)
+X		fprintf(stderr, "error: device configuration failed\n");
+X	
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' &&
+  chmod 0664 'MultiSound.d/pinnaclecfg.c' ||
+  $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed'
+366bdf27f0db767a3c7921d0a6db20fe  MultiSound.d/pinnaclecfg.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`"
+    test 10235 -eq "$shar_count" ||
+    $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10235,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/Makefile ==============
+if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' &&
+CC	= gcc
+CFLAGS	= -O
+PROGS	= setdigital msndreset pinnaclecfg conv
+X
+all: $(PROGS)
+X
+clean:
+X	rm -f $(PROGS)
+SHAR_EOF
+  $shar_touch -am 1204092398 'MultiSound.d/Makefile' &&
+  chmod 0664 'MultiSound.d/Makefile' ||
+  $echo 'restore of' 'MultiSound.d/Makefile' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/Makefile:' 'MD5 check failed'
+76ca8bb44e3882edcf79c97df6c81845  MultiSound.d/Makefile
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`"
+    test 106 -eq "$shar_count" ||
+    $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/conv.l ==============
+if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' &&
+%%
+[ \n\t,\r]
+\;.*
+DB
+[0-9A-Fa-f]+H	{ int n; sscanf(yytext, "%xH", &n); printf("%c", n); }
+%%
+int yywrap() { return 1; }
+main() { yylex(); }
+SHAR_EOF
+  $shar_touch -am 0828231798 'MultiSound.d/conv.l' &&
+  chmod 0664 'MultiSound.d/conv.l' ||
+  $echo 'restore of' 'MultiSound.d/conv.l' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/conv.l:' 'MD5 check failed'
+d2411fc32cd71a00dcdc1f009e858dd2  MultiSound.d/conv.l
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`"
+    test 141 -eq "$shar_count" ||
+    $echo 'MultiSound.d/conv.l:' 'original size' '141,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/msndreset.c ==============
+if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' &&
+/*********************************************************************
+X *
+X * msndreset.c - resets the MultiSound card
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+#include 
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: msndreset \n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) {
+X		fprintf(stderr, "error: msnd ioctl reset failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X	
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' &&
+  chmod 0664 'MultiSound.d/msndreset.c' ||
+  $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed'
+c52f876521084e8eb25e12e01dcccb8a  MultiSound.d/msndreset.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`"
+    test 1472 -eq "$shar_count" ||
+    $echo 'MultiSound.d/msndreset.c:' 'original size' '1472,' 'current size' "$shar_count!"
+  fi
+fi
+rm -fr _sh01426
+exit 0
diff --git a/Documentation/sound/oss/OPL3 b/Documentation/sound/oss/OPL3
new file mode 100644
index 00000000..2468ff82
--- /dev/null
+++ b/Documentation/sound/oss/OPL3
@@ -0,0 +1,6 @@
+A pure OPL3 card is nice and easy to configure. Simply do
+
+insmod opl3 io=0x388
+
+Change the I/O address in the very unlikely case this card is differently
+configured
diff --git a/Documentation/sound/oss/Opti b/Documentation/sound/oss/Opti
new file mode 100644
index 00000000..c15af3c0
--- /dev/null
+++ b/Documentation/sound/oss/Opti
@@ -0,0 +1,222 @@
+Support for the OPTi 82C931 chip
+--------------------------------
+Note: parts of this README file apply also to other
+cards that use the mad16 driver.
+
+Some items in this README file are based on features
+added to the sound driver after Linux-2.1.91 was out.
+By the time of writing this I do not know which official
+kernel release will include these features.
+Please do not report inconsistencies on older Linux
+kernels.
+
+The OPTi 82C931 is supported in its non-PnP mode.
+Usually you do not need to set jumpers, etc. The sound driver
+will check the card status and if it is required it will
+force the card into a mode in which it can be programmed.
+
+If you have another OS installed on your computer it is recommended
+that Linux and the other OS use the same resources.
+
+Also, it is recommended that resources specified in /etc/modprobe.conf
+and resources specified in /etc/isapnp.conf agree.
+
+Compiling the sound driver
+--------------------------
+I highly recommend that you build a modularized sound driver.
+This document does not cover a sound-driver which is built in
+the kernel.
+
+Sound card support should be enabled as a module (chose m).
+Answer 'm' for  these items:
+  Generic OPL2/OPL3 FM synthesizer support		(CONFIG_SOUND_ADLIB)
+  Microsoft Sound System support			(CONFIG_SOUND_MSS)
+  Support for OPTi MAD16 and/or Mozart based cards	(CONFIG_SOUND_MAD16)
+  FM synthesizer (YM3812/OPL-3) support			(CONFIG_SOUND_YM3812)
+
+The configuration menu may ask for addresses, IRQ lines or DMA
+channels. If the card is used as a module the module loading
+options will override these values.
+
+For the OPTi 931 you can answer 'n' to:
+  Support MIDI in older MAD16 based cards (requires SB)	(CONFIG_SOUND_MAD16_OLDCARD)
+If you do need MIDI support in a Mozart or C928 based card you
+need to answer 'm' to the above question. In that case you will
+also need to answer 'm' to:
+  '100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support' (CONFIG_SOUND_SB)
+
+Go on and compile your kernel and modules. Install the modules. Run depmod -a.
+
+Using isapnptools
+-----------------
+In most systems with a PnP BIOS you do not need to use isapnp. The
+initialization provided by the BIOS is sufficient for the driver
+to pick up the card and continue initialization.
+
+If that fails, or if you have other PnP cards, you need to use isapnp
+to initialize the card.
+This was tested with isapnptools-1.11 but I recommend that you use
+isapnptools-1.13 (or newer). Run pnpdump to dump the information
+about your PnP cards. Then edit the resulting file and select
+the options of your choice. This file is normally installed as
+/etc/isapnp.conf.
+
+The driver has one limitation with respect to I/O port resources:
+IO3 base must be 0x0E0C.  Although isapnp allows other ports, this
+address is hard-coded into the driver.
+
+Using kmod and autoloading the sound driver
+-------------------------------------------
+Comment: as of linux-2.1.90 kmod is replacing kerneld.
+The config file '/etc/modprobe.conf' is used as before.
+
+This is the sound part of my /etc/modprobe.conf file.
+Following that I will explain each line.
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0  mad16
+alias synth0 opl3
+options sb mad16=1
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+install mad16 /sbin/modprobe -i mad16 && /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+If you have an MPU daughtercard or onboard MPU you will want to add to the
+"options mad16" line - eg 
+
+options mad16 irq=5 dma=0 dma16=3 io=0x530 mpu_io=0x330 mpu_irq=9
+
+To set the I/O and IRQ of the MPU.
+
+
+Explain:
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0  mad16
+alias synth0 opl3
+
+When any sound device is opened the kernel requests auto-loading
+of char-major-14. There is a built-in alias that translates this
+request to loading the main sound module.
+
+The sound module in its turn will request loading of a sub-driver
+for mixer, audio, midi or synthesizer device. The first 3 are
+supported by the mad16 driver. The synth device is supported
+by the opl3 driver.
+
+There is currently no way to autoload the sound device driver
+if more than one card is installed.
+
+options sb mad16=1
+
+This is left for historical reasons. If you enable the
+config option 'Support MIDI in older MAD16 based cards (requires SB)'
+or if you use an older mad16 driver it will force loading of the
+SoundBlaster driver. This option tells the SB driver not to look
+for a SB card but to wait for the mad16 driver.
+
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+
+post-install mad16 /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+This sets resources and options for the mad16 and opl3 drivers.
+I use two DMA channels (only one is required) to enable full duplex.
+joystick=1 enables the joystick port. cdtype=0 disables the cd port.
+You can also set mpu_io and mpu_irq in the mad16 options for the
+uart401 driver.
+
+This tells modprobe to run /sbin/ad1848_mixer_reroute after
+mad16 is successfully loaded and initialized. The source
+for ad1848_mixer_reroute is appended to the end of this readme
+file. It is impossible for the sound driver to know the actual
+connections to the mixer. The 3 inputs intended for cd, synth
+and line-in are mapped to the generic inputs line1, line2 and
+line3. This program reroutes these mixer channels to their
+right names (note the right mapping depends on the actual sound
+card that you use).
+The numeric parameters mean:
+ 14=line1 8=cd    - reroute line1 to the CD input.
+ 15=line2 3=synth - reroute line2 to the synthesizer input.
+ 16=line3 6=line  - reroute line3 to the line input.
+For reference on other input names look at the file
+/usr/include/linux/soundcard.h.
+
+Using a joystick
+-----------------
+You must enable a joystick in the mad16 options. (also
+in /etc/isapnp.conf if you use it).
+Tested with regular analog joysticks.
+
+A CDROM drive connected to the sound card
+-----------------------------------------
+The 82C931 chip has support only for secondary ATAPI cdrom.
+(cdtype=8). Loading the mad16 driver resets the C931 chip
+and if a cdrom was already mounted it may cause a complete
+system hang. Do not use the sound card if you have an alternative.
+If you do use the sound card it is important that you load
+the mad16 driver (use "modprobe mad16" to prevent auto-unloading)
+before the cdrom is accessed the first time.
+
+Using the sound driver built-in to the kernel may help here, but...
+Most new systems have a PnP BIOS and also two IDE controllers.
+The IDE controller on the sound card may be needed only on older
+systems (which have only one IDE controller) but these systems
+also do not have a PnP BIOS - requiring isapnptools and a modularized
+driver.
+
+Known problems
+--------------
+1. See the section on "A CDROM drive connected to the sound card".
+
+2. On my system the codec cannot capture companded sound samples.
+   (eg., recording from /dev/audio). When any companded capture is
+   requested I get stereo-16 bit samples instead. Playback of
+   companded samples works well. Apparently this problem is not common
+   to all C931 based cards. I do not know how to identify cards that
+   have this problem.
+
+Source for ad1848_mixer_reroute.c
+---------------------------------
+#include 
+#include 
+#include 
+
+static char *mixer_names[SOUND_MIXER_NRDEVICES] =
+	SOUND_DEVICE_LABELS;
+
+int
+main(int argc, char **argv) {
+	int val, from, to;
+	int i, fd;
+
+	fd = open("/dev/mixer", O_RDWR);
+	if(fd < 0) {
+		perror("/dev/mixer");
+		return 1;
+	}
+
+	for(i = 2; i < argc; i += 2) {
+		from = atoi(argv[i-1]);
+		to = atoi(argv[i]);
+
+		if(to == SOUND_MIXER_NONE)
+			fprintf(stderr, "%s: turning off mixer %s\n",
+				argv[0], mixer_names[to]);
+		else
+			fprintf(stderr, "%s: rerouting mixer %s to %s\n",
+				argv[0], mixer_names[from], mixer_names[to]);
+
+		val = from << 8 | to;
+
+		if(ioctl(fd, SOUND_MIXER_PRIVATE2, &val)) {
+			perror("AD1848 mixer reroute");
+			return 1;
+		}
+	}
+
+	return 0;
+}
+
diff --git a/Documentation/sound/oss/PAS16 b/Documentation/sound/oss/PAS16
new file mode 100644
index 00000000..951b3dce
--- /dev/null
+++ b/Documentation/sound/oss/PAS16
@@ -0,0 +1,163 @@
+Pro Audio Spectrum 16 for 2.3.99 and later
+=========================================
+by Thomas Molina (tmolina@home.com)
+last modified 3 Mar 2001
+Acknowledgement to Axel Boldt (boldt@math.ucsb.edu) for stuff taken
+from Configure.help, Riccardo Facchetti for stuff from README.OSS,
+and others whose names I could not find.
+
+This documentation is relevant for the PAS16 driver (pas2_card.c and
+friends) under kernel version 2.3.99 and later.  If you are
+unfamiliar with configuring sound under Linux, please read the
+Sound-HOWTO, Documentation/sound/oss/Introduction and other
+relevant docs first.
+
+The following information is relevant information from README.OSS
+and legacy docs for the Pro Audio Spectrum 16 (PAS16):
+==================================================================
+
+The pas2_card.c driver supports the following cards --
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+        Pro Audio Spectrum 16
+        Pro Audio Studio 16
+        Logitech Sound Man 16
+        NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+              and will not be supported by the driver.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the question about (PAS16) if you don't really have a PAS16.  Sound
+configuration may also be made modular by answering m to configuration
+options presented. 
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+  "ProAudioSpectrum 16 support",
+        - Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+          Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+          you read the above list correctly). Don't answer 'y' if you
+          have some other card made by Media Vision or Logitech since they
+          are not PAS16 compatible.
+          NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+          if you want to use the SB emulation of PAS16. It's also possible to
+          the emulation if you want to use a true SB card together with PAS16
+          (there is another question about this that is asked later).
+
+  "Generic OPL2/OPL3 FM synthesizer support",
+        - Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+	  The PAS16 has an OPL3-compatible FM chip.
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+
+The new stuff for 2.3.99 and later
+============================================================================
+The following configuration options from Documentation/Configure.help
+are relevant to configuring the PAS16:
+
+Sound card support
+CONFIG_SOUND
+  If you have a sound card in your computer, i.e. if it can say more
+  than an occasional beep, say Y. Be sure to have all the information
+  about your sound card and its configuration down (I/O port,
+  interrupt and DMA channel), because you will be asked for it.
+
+  You want to read the Sound-HOWTO, available from
+  http://www.tldp.org/docs.html#howto . General information
+  about the modular sound system is contained in the files
+  Documentation/sound/oss/Introduction. The file
+  Documentation/sound/oss/README.OSS contains some slightly outdated but
+  still useful information as well.
+
+OSS sound modules
+CONFIG_SOUND_OSS
+  OSS is the Open Sound System suite of sound card drivers. They make
+  sound programming easier since they provide a common API. Say Y or M
+  here (the module will be called sound.o) if you haven't found a
+  driver for your sound card above, then pick your driver from the
+  list below.
+
+Persistent DMA buffers
+CONFIG_SOUND_DMAP
+  Linux can often have problems allocating DMA buffers for ISA sound
+  cards on machines with more than 16MB of RAM. This is because ISA
+  DMA buffers must exist below the 16MB boundary and it is quite
+  possible that a large enough free block in this region cannot be
+  found after the machine has been running for a while. If you say Y
+  here the DMA buffers (64Kb) will be allocated at boot time and kept
+  until the shutdown. This option is only useful if you said Y to
+  "OSS sound modules", above. If you said M to "OSS sound modules"
+  then you can get the persistent DMA buffer functionality by passing
+  the command-line argument "dmabuf=1" to the sound.o module.
+
+  Say y here for PAS16.
+
+ProAudioSpectrum 16 support
+CONFIG_SOUND_PAS
+  Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
+  16 or Logitech SoundMan 16 sound card. Don't answer Y if you have
+  some other card made by Media Vision or Logitech since they are not
+  PAS16 compatible.  It is not necessary to enable the  separate
+  Sound Blaster support; it is included in the PAS driver.
+
+  If you compile the driver into the kernel, you have to add
+  "pas2=,,,,,,,
+  to the kernel command line.
+
+FM Synthesizer (YM3812/OPL-3) support
+CONFIG_SOUND_YM3812
+  Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+  Answering Y is usually a safe and recommended choice, however some
+  cards may have software (TSR) FM emulation. Enabling FM support with
+  these cards may cause trouble (I don't currently know of any such
+  cards, however).
+  Please read the file Documentation/sound/oss/OPL3 if your card has an
+  OPL3 chip.
+  If you compile the driver into the kernel, you have to add
+  "opl3=" to the kernel command line.
+
+  If you compile your drivers into the kernel, you MUST configure 
+  OPL3 support as a module for PAS16 support to work properly.
+  You can then get OPL3 functionality by issuing the command:
+  insmod opl3
+  In addition, you must either add the following line to 
+  /etc/modprobe.conf:
+  options opl3 io=0x388
+  or else add the following line to /etc/lilo.conf:
+  opl3=0x388
+
+
+EXAMPLES
+===================================================================
+To use the PAS16 in my computer I have enabled the following sound
+configuration options:
+
+CONFIG_SOUND=y
+CONFIG_SOUND_OSS=y
+CONFIG_SOUND_TRACEINIT=y
+CONFIG_SOUND_DMAP=y
+CONFIG_SOUND_PAS=y
+CONFIG_SOUND_SB=n
+CONFIG_SOUND_YM3812=m
+
+I have also included the following append line in /etc/lilo.conf:
+append="pas2=0x388,10,3,-1,0x220,5,1,-1 sb=0x220,5,1,-1 opl3=0x388"
+
+The io address of 0x388 is default configuration on the PAS16.  The
+irq of 10 and dma of 3 may not match your installation.  The above 
+configuration enables PAS16, 8-bit Soundblaster and OPL3
+functionality.  If Soundblaster functionality is not desired, the
+following line would be appropriate:
+append="pas2=0x388,10,3,-1,0,-1,-1,-1 opl3=0x388"
+
+If sound is built totally modular, the above options may be 
+specified in /etc/modprobe.conf for pas2, sb and opl3
+respectively. 
diff --git a/Documentation/sound/oss/PSS b/Documentation/sound/oss/PSS
new file mode 100644
index 00000000..187b9525
--- /dev/null
+++ b/Documentation/sound/oss/PSS
@@ -0,0 +1,41 @@
+The PSS cards and other ECHO based cards provide an onboard DSP with 
+downloadable programs and also has an AD1848 "Microsoft Sound System" 
+device. The PSS driver enables MSS and MPU401 modes of the card. SB 
+is not enabled since it doesn't work concurrently with MSS. 
+
+If you build this driver as a module then the driver takes the following
+parameters
+
+pss_io. 	The I/O base the PSS card is configured at (normally 0x220
+		or 0x240)
+
+mss_io		The base address of the Microsoft Sound System interface.
+		This is normally 0x530, but may be 0x604 or other addresses.
+
+mss_irq		The interrupt assigned to the Microsoft Sound System
+		emulation. IRQ's 3,5,7,9,10,11 and 12 are available. If you
+		get IRQ errors be sure to check the interrupt is set to 
+		"ISA/Legacy" in the BIOS on modern machines.
+
+mss_dma		The DMA channel used by the Microsoft Sound System.
+		This can be 0, 1, or 3. DMA 0 is not available on older 
+		machines and will cause a crash on them.
+
+mpu_io		The MPU emulation base address. This sets the base of the
+		synthesizer. It is typically 0x330 but can be altered.
+
+mpu_irq		The interrupt to use for the synthesizer. It must differ
+		from the IRQ used by the Microsoft Sound System port.
+
+
+The mpu_io/mpu_irq fields are optional. If they are not specified the 
+synthesizer parts are not configured.
+
+When the module is loaded it looks for a file called 
+/etc/sound/pss_synth. This is the firmware file from the DOS install disks.
+This fil holds a general MIDI emulation. The file expected is called
+genmidi.ld on newer DOS driver install disks and synth.ld on older ones.
+
+You can also load alternative DSP algorithms into the card if you wish. One
+alternative driver can be found at http://www.mpg123.de/
+
diff --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates
new file mode 100644
index 00000000..c84dd759
--- /dev/null
+++ b/Documentation/sound/oss/PSS-updates
@@ -0,0 +1,88 @@
+	This file contains notes for users of PSS sound cards who wish to use the 
+newly added features of the newest version of this driver.
+
+	The major enhancements present in this new revision of this driver is the 
+addition of two new module parameters that allow you to take full advantage of 
+all the features present on your PSS sound card.  These features include the 
+ability to enable both the builtin CDROM and joystick ports.
+
+pss_enable_joystick
+
+	This parameter is basically a flag.  A 0 will leave the joystick port 
+disabled, while a non-zero value would enable the joystick port.  The default 
+setting is pss_enable_joystick=0 as this keeps this driver fully compatible 
+with systems that were using previous versions of this driver.  If you wish to 
+enable the joystick port you will have to add pss_enable_joystick=1 as an 
+argument to the driver.  To actually use the joystick port you will then have 
+to load the joystick driver itself.  Just remember to load the joystick driver 
+AFTER the pss sound driver.
+
+pss_cdrom_port
+
+	This parameter takes a port address as its parameter.  Any available port 
+address can be specified to enable the CDROM port, except for 0x0 and -1 as 
+these values would leave the port disabled.  Like the joystick port, the cdrom 
+port will require that an appropriate CDROM driver be loaded before you can make 
+use of the newly enabled CDROM port.  Like the joystick port option above, 
+remember to load the CDROM driver AFTER the pss sound driver.  While it may 
+differ on some PSS sound cards, all the PSS sound cards that I have seen have a 
+builtin Wearnes CDROM port.  If this is the case with your PSS sound card you 
+should load aztcd with the appropriate port option that matches the port you 
+assigned to the CDROM port when you loaded your pss sound driver.  (ex. 
+modprobe pss pss_cdrom_port=0x340 && modprobe aztcd aztcd=0x340)  The default 
+setting of this parameter leaves the CDROM port disabled to maintain full 
+compatibility with systems using previous versions of this driver.
+
+	Other options have also been added for the added convenience and utility 
+of the user.  These options are only available if this driver is loaded as a 
+module.
+
+pss_no_sound
+
+	This module parameter is a flag that can be used to tell the driver to 
+just configure non-sound components.  0 configures all components, a non-0 
+value will only attept to configure the CDROM and joystick ports.  This 
+parameter can be used by a user who only wished to use the builtin joystick 
+and/or CDROM port(s) of his PSS sound card.  If this driver is loaded with this 
+parameter and with the parameter below set to true then a user can safely unload 
+this driver with the following command "rmmod pss && rmmod ad1848 && rmmod 
+mpu401 && rmmod sound && rmmod soundcore" and retain the full functionality of 
+his CDROM and/or joystick port(s) while gaining back the memory previously used 
+by the sound drivers.  This default setting of this parameter is 0 to retain 
+full behavioral compatibility with previous versions of this driver.
+
+pss_keep_settings
+
+	This parameter can be used to specify whether you want the driver to reset 
+all emulations whenever its unloaded.  This can be useful for those who are 
+sharing resources (io ports, IRQ's, DMA's) between different ISA cards.  This 
+flag can also be useful in that future versions of this driver may reset all 
+emulations by default on the driver's unloading (as it probably should), so 
+specifying it now will ensure that all future versions of this driver will 
+continue to work as expected.  The default value of this parameter is 1 to 
+retain full behavioral compatibility with previous versions of this driver.
+
+pss_firmware
+
+	This parameter can be used to specify the file containing the firmware 
+code so that a user could tell the driver where that file is located instead 
+of having to put it in a predefined location with a predefined name.  The 
+default setting of this parameter is "/etc/sound/pss_synth" as this was the 
+path and filename the hardcoded value in the previous versions of this driver.
+
+Examples:
+
+# Normal PSS sound card system, loading of drivers.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 mpu_io=0x338 mpu_irq=9 mss_io=0x530 mss_irq=10 mss_dma=1 pss_cdrom_port=0x340 pss_enable_joystick=1
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
+
+# System using the PSS sound card just for its CDROM and joystick ports.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 pss_cdrom_port=0x340 pss_enable_joystick=1 pss_no_sound=1
+/sbin/rmmod pss && /sbin/rmmod ad1848 && /sbin/rmmod mpu401 && /sbin/rmmod sound && /sbin/rmmod soundcore  # This line not needed, but saves memory.
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS
new file mode 100644
index 00000000..4be25942
--- /dev/null
+++ b/Documentation/sound/oss/README.OSS
@@ -0,0 +1,1455 @@
+Introduction
+------------
+
+This file is a collection of all the old Readme files distributed with
+OSS/Lite by Hannu Savolainen. Since the new Linux sound driver is founded
+on it I think these information may still be interesting for users that
+have to configure their sound system.
+
+Be warned: Alan Cox is the current maintainer of the Linux sound driver so if
+you have problems with it, please contact him or the current device-specific
+driver maintainer (e.g. for aedsp16 specific problems contact me). If you have
+patches, contributions or suggestions send them to Alan: I'm sure they are
+welcome.
+
+In this document you will find a lot of references about OSS/Lite or ossfree:
+they are gone forever. Keeping this in mind and with a grain of salt this
+document can be still interesting and very helpful.
+
+[ File edited 17.01.1999 - Riccardo Facchetti ]
+[ Edited miroSOUND section 19.04.2001 - Robert Siemer ]
+
+OSS/Free version 3.8 release notes
+----------------------------------
+
+Please read the SOUND-HOWTO (available from sunsite.unc.edu and other Linux FTP 
+sites). It gives instructions about using sound with Linux. It's bit out of
+date but still very useful. Information about bug fixes and such things
+is available from the web page (see above).
+
+Please check http://www.opensound.com/pguide for more info about programming
+with OSS API.
+
+   ====================================================
+-  THIS VERSION ____REQUIRES____ Linux 2.1.57 OR LATER.
+   ====================================================
+
+Packages "snd-util-3.8.tar.gz" and "snd-data-0.1.tar.Z"
+contain useful utilities to be used with this driver.
+See http://www.opensound.com/ossfree/ for
+download instructions.
+
+If you are looking for the installation instructions, please
+look forward into this document.
+
+Supported sound cards
+---------------------
+
+See below.
+
+Contributors
+------------
+
+This driver contains code by several contributors. In addition several other
+persons have given useful suggestions. The following is a list of major
+contributors. (I could have forgotten some names.)
+
+	Craig Metz	1/2 of the PAS16 Mixer and PCM support
+	Rob Hooft	Volume computation algorithm for the FM synth.
+	Mika Liljeberg	uLaw encoding and decoding routines
+	Jeff Tranter	Linux SOUND HOWTO document
+	Greg Lee	Volume computation algorithm for the GUS and
+			lots of valuable suggestions.
+	Andy Warner	ISC port
+	Jim Lowe,
+	Amancio Hasty Jr	FreeBSD/NetBSD port
+	Anders Baekgaard 	Bug hunting and valuable suggestions.
+	Joerg Schubert	SB16 DSP support (initial version).
+	Andrew Robinson Improvements to the GUS driver
+	Megens SA	MIDI recording for SB and SB Pro (initial version).
+	Mikael Nordqvist  Linear volume support for GUS and
+			  nonblocking /dev/sequencer.
+	Ian Hartas		SVR4.2 port
+	Markus Aroharju	and
+	Risto Kankkunen		Major contributions to the mixer support
+				of GUS v3.7.
+	Hunyue Yau	Mixer support for SG NX Pro.
+	Marc Hoffman	PSS support (initial version).
+	Rainer Vranken	Initialization for Jazz16 (initial version).
+	Peter Trattler	Initial version of loadable module support for Linux.
+	JRA Gibson	16 bit mode for Jazz16 (initial version)
+	Davor Jadrijevic MAD16 support (initial version)
+	Gregor Hoffleit	Mozart support (initial version)
+	Riccardo Facchetti Audio Excel DSP 16 (aedsp16) support
+	James Hightower Spotting a tiny but important bug in CS423x support.
+	Denis Sablic	OPTi 82C924 specific enhancements (non PnP mode)
+	Tim MacKenzie	Full duplex support for OPTi 82C930.
+	
+	Please look at lowlevel/README for more contributors.
+
+There are probably many other names missing. If you have sent me some
+patches and your name is not in the above list, please inform me.
+
+Sending your contributions or patches
+-------------------------------------
+
+First of all it's highly recommended to contact me before sending anything
+or before even starting to do any work. Tell me what you suggest to be
+changed or what you have planned to do. Also ensure you are using the
+very latest (development) version of OSS/Free since the change may already be
+implemented there. In general it's a major waste of time to try to improve a
+several months old version. Information about the latest version can be found
+from http://www.opensound.com/ossfree. In general there is no point in
+sending me patches relative to production kernels.
+
+Sponsors etc.
+-------------
+
+The following companies have greatly helped development of this driver 
+in form of a free copy of their product:
+
+Novell, Inc.		UnixWare personal edition + SDK
+The Santa Cruz Operation, Inc. 	A SCO OpenServer + SDK
+Ensoniq Corp,		a SoundScape card and extensive amount of assistance
+MediaTrix Peripherals Inc, a AudioTrix Pro card + SDK
+Acer, Inc.		a pair of AcerMagic S23 cards.
+
+In addition the following companies have provided me sufficient amount
+of technical information at least some of their products (free or $$$):
+
+Advanced Gravis Computer Technology Ltd.
+Media Vision Inc.
+Analog Devices Inc.
+Logitech Inc.
+Aztech Labs Inc.
+Crystal Semiconductor Corporation,
+Integrated Circuit Systems Inc.
+OAK Technology
+OPTi
+Turtle Beach
+miro
+Ad Lib Inc. ($$)
+Music Quest Inc. ($$)
+Creative Labs ($$$)
+
+If you have some problems
+=========================
+
+Read the sound HOWTO (sunsite.unc.edu:/pub/Linux/docs/...?).
+Also look at the home page (http://www.opensound.com/ossfree). It may
+contain info about some recent bug fixes.
+
+It's likely that you have some problems when trying to use the sound driver
+first time. Sound cards don't have standard configuration so there are no
+good default configuration to use. Please try to use same I/O, DMA and IRQ
+values for the sound card than with DOS.
+
+If you get an error message when trying to use the driver, please look
+at /var/adm/messages for more verbose error message.
+
+
+The following errors are likely with /dev/dsp and /dev/audio.
+
+	- "No such device or address".
+	This error indicates that there are no suitable hardware for the
+	device file or the sound driver has been compiled without support for
+	this particular device. For example /dev/audio and /dev/dsp will not
+	work if "digitized voice support" was not enabled during "make config".
+	
+	- "Device or resource busy". Probably the IRQ (or DMA) channel 
+	required by the sound card is in use by some other device/driver.
+
+	- "I/O error". Almost certainly (99%) it's an IRQ or DMA conflict.
+	Look at the kernel messages in /var/adm/notice for more info.
+
+	- "Invalid argument". The application is calling ioctl()
+	with impossible parameters. Check that the application is
+	for sound driver version 2.X or later.
+
+Linux installation
+==================
+
+IMPORTANT!	Read this if you are installing a separately
+		distributed version of this driver.
+
+		Check that your kernel version works with this
+		release of the driver (see Readme). Also verify
+		that your current kernel version doesn't have more
+		recent sound driver version than this one. IT'S HIGHLY
+		RECOMMENDED THAT YOU USE THE SOUND DRIVER VERSION THAT
+		IS DISTRIBUTED WITH KERNEL SOURCES.
+
+- When installing separately distributed sound driver you should first
+  read the above notice. Then try to find proper directory where and how
+  to install the driver sources. You should not try to install a separately
+  distributed driver version if you are not able to find the proper way
+  yourself (in this case use the version that is distributed with kernel
+  sources). Remove old version of linux/drivers/sound directory before
+  installing new files.
+
+- To build the device files you need to run the enclosed shell script 
+  (see below). You need to do this only when installing sound driver
+  first time or when upgrading to much recent version than the earlier
+  one.
+
+- Configure and compile Linux as normally (remember to include the
+  sound support during "make config"). Please refer to kernel documentation
+  for instructions about configuring and compiling kernel. File Readme.cards
+  contains card specific instructions for configuring this driver for
+  use with various sound cards.
+
+Boot time configuration (using lilo and insmod) 
+-----------------------------------------------
+
+This information has been removed. Too many users didn't believe
+that it's really not necessary to use this method. Please look at
+Readme of sound driver version 3.0.1 if you still want to use this method.
+
+Problems
+--------
+
+Common error messages:
+
+- /dev/???????: No such file or directory.
+Run the script at the end of this file.
+
+- /dev/???????: No such device.
+You are not running kernel which contains the sound driver. When using
+modularized sound driver this error means that the sound driver is not
+loaded.
+
+- /dev/????: No such device or address.
+Sound driver didn't detect suitable card when initializing. Please look at
+Readme.cards for info about configuring the driver with your card. Also
+check for possible boot (insmod) time error messages in /var/adm/messages.
+
+- Other messages or problems
+Please check http://www.opensound.com/ossfree for more info.
+
+Configuring version 3.8 (for Linux) with some common sound cards
+================================================================
+
+This document describes configuring sound cards with the freeware version of
+Open Sound Systems (OSS/Free). Information about the commercial version
+(OSS/Linux) and its configuration is available from 
+http://www.opensound.com/linux.html. Information presented here is
+not valid for OSS/Linux. 
+
+If you are unsure about how to configure OSS/Free
+you can download the free evaluation version of OSS/Linux from the above
+address. There is a chance that it can autodetect your sound card. In this case
+you can use the information included in soundon.log when configuring OSS/Free.
+
+
+IMPORTANT!	This document covers only cards that were "known" when
+		this driver version was released. Please look at
+		http://www.opensound.com/ossfree for info about
+		cards introduced recently.
+
+		When configuring the sound driver, you should carefully
+		check each sound configuration option (particularly
+		"Support for /dev/dsp and /dev/audio"). The default values
+		offered by these programs are not necessarily valid.
+
+
+THE BIGGEST MISTAKES YOU CAN MAKE
+=================================
+
+1. Assuming that the card is Sound Blaster compatible when it's not.
+--------------------------------------------------------------------
+
+The number one mistake is to assume that your card is compatible with
+Sound Blaster. Only the cards made by Creative Technology or which have
+one or more chips labeled by Creative are SB compatible. In addition there
+are few sound chipsets which are SB compatible in Linux such as ESS1688 or
+Jazz16. Note that SB compatibility in DOS/Windows does _NOT_ mean anything
+in Linux. 
+
+IF YOU REALLY ARE 150% SURE YOU HAVE A SOUND BLASTER YOU CAN SKIP THE REST OF
+THIS CHAPTER.
+
+For most other "supposed to be SB compatible" cards you have to use other
+than SB drivers (see below).  It is possible to get most sound cards to work
+in SB mode but in general it's a complete waste of time. There are several
+problems which you will encounter by using SB mode with cards that are not
+truly SB compatible:
+
+- The SB emulation is at most SB Pro (DSP version 3.x) which means that 
+you get only 8 bit audio (there is always an another ("native") mode which
+gives the 16 bit capability). The 8 bit only operation is the reason why
+many users claim that sound quality in Linux is much worse than in DOS.
+In addition some applications require 16 bit mode and they produce just
+noise with a 8 bit only device.
+- The card may work only in some cases but refuse to work most of the
+time. The SB compatible mode always requires special initialization which is 
+done by the DOS/Windows drivers. This kind of cards work in Linux after
+you have warm booted it after DOS but they don't work after cold boot
+(power on or reset).
+- You get the famous "DMA timed out" messages. Usually all SB clones have
+software selectable IRQ and DMA settings. If the (power on default) values
+currently used by the card don't match configuration of the driver you will
+get the above error message whenever you try to record or play. There are
+few other reasons to the DMA timeout message but using the SB mode seems
+to be the most common cause.
+
+2. Trying to use a PnP (Plug & Play) card just like an ordinary sound card
+--------------------------------------------------------------------------
+
+Plug & Play is a protocol defined by Intel and Microsoft. It lets operating
+systems to easily identify and reconfigure I/O ports, IRQs and DMAs of ISA
+cards. The problem with PnP cards is that the standard Linux doesn't currently
+(versions 2.1.x and earlier) don't support PnP. This means that you will have
+to use some special tricks (see later) to get a PnP card alive. Many PnP cards
+work after they have been initialized but this is not always the case.
+
+There are sometimes both PnP and non-PnP versions of the same sound card.
+The non-PnP version is the original model which usually has been discontinued
+more than an year ago. The PnP version has the same name but with "PnP"
+appended to it (sometimes not). This causes major confusion since the non-PnP
+model works with Linux but the PnP one doesn't.
+
+You should carefully check if "Plug & Play" or "PnP" is mentioned in the name
+of the card or in the documentation or package that came with the card. 
+Everything described in the rest of this document is not necessarily valid for
+PnP models of sound cards even you have managed to wake up the card properly.
+Many PnP cards are simply too different from their non-PnP ancestors which are
+covered by this document.
+
+
+Cards that are not (fully) supported by this driver
+===================================================
+
+See http://www.opensound.com/ossfree for information about sound cards 
+to be supported in future. 
+
+
+How to use sound without recompiling kernel and/or sound driver
+===============================================================
+
+There is a commercial sound driver which comes in precompiled form and doesn't
+require recompiling of the kernel. See http://www.4Front-tech.com/oss.html for
+more info.
+
+
+Configuring PnP cards
+=====================
+
+New versions of most sound cards use the so-called ISA PnP protocol for
+soft configuring their I/O, IRQ, DMA and shared memory resources.
+Currently at least cards made by Creative Technology (SB32 and SB32AWE
+PnP), Gravis (GUS PnP and GUS PnP Pro), Ensoniq (Soundscape PnP) and
+Aztech (some Sound Galaxy models) use PnP technology. The CS4232/4236 audio
+chip by Crystal Semiconductor (Intel Atlantis, HP Pavilion and many other
+motherboards) is also based on PnP technology but there is a "native" driver
+available for it (see information about CS4232 later in this document).
+
+PnP sound cards (as well as most other PnP ISA cards) are not supported
+by this version of the driver . Proper
+support for them should be released during 97 once the kernel level
+PnP support is available.
+
+There is a method to get most of the PnP cards to work. The basic method
+is the following:
+
+1) Boot DOS so the card's DOS drivers have a chance to initialize it.
+2) _Cold_ boot to Linux by using "loadlin.exe".  Hitting ctrl-alt-del
+works with older machines but causes a hard reset of all cards on recent
+(Pentium) machines.
+3) If you have the sound driver in Linux configured properly, the card should
+work now. "Proper" means that I/O, IRQ and DMA settings are the same as in
+DOS. The hard part is to find which settings were used. See the documentation of
+your card for more info.
+
+Windows 95 could work as well as DOS but running loadlin may be difficult.
+Probably you should "shut down" your machine to MS-DOS mode before running it.
+
+Some machines have a BIOS utility for setting PnP resources. This is a good
+way to configure some cards. In this case you don't need to boot DOS/Win95
+before starting Linux.
+
+Another way to initialize PnP cards without DOS/Win95 is a Linux based
+PnP isolation tool. When writing this there is a pre alpha test version
+of such a tool available from ftp://ftp.demon.co.uk/pub/unix/linux/utils. The
+file is called isapnptools-*. Please note that this tool is just a temporary
+solution which may be incompatible with future kernel versions having proper
+support for PnP cards. There are bugs in setting DMA channels in earlier
+versions of isapnptools so at least version 1.6 is required with sound cards.
+
+Yet another way to use PnP cards is to use (commercial) OSS/Linux drivers.  See
+http://www.opensound.com/linux.html for more info. This is probably the way you
+should do it if you don't want to spend time recompiling the kernel and 
+required tools.
+
+
+Read this before trying to configure the driver
+===============================================
+
+There are currently many cards that work with this driver. Some of the cards
+have native support while others work since they emulate some other
+card (usually SB, MSS/WSS and/or MPU401). The following cards have native
+support in the driver. Detailed instructions for configuring these cards
+will be given later in this document.
+
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+	Pro Audio Spectrum 16
+	Pro Audio Studio 16
+	Logitech Sound Man 16
+	NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+	      and will not be supported by the driver.
+
+Media Vision Jazz16 based cards
+	Pro Sonic 16
+	Logitech SoundMan Wave
+	(Other Jazz based cards should work but I don't have any reports
+	about them).
+
+Sound Blasters
+	SB 1.0 to 2.0
+	SB Pro
+	SB 16
+	SB32/64/AWE
+		Configure SB32/64/AWE just like SB16. See lowlevel/README.awe
+		for information about using the wave table synth.
+	        NOTE! AWE63/Gold and 16/32/AWE "PnP" cards need to be activated
+		      using isapnptools before they work with OSS/Free.
+	SB16 compatible cards by other manufacturers than Creative.
+		You have been fooled since there are _no_ SB16 compatible
+		cards on the market (as of May 1997). It's likely that your card
+		is compatible just with SB Pro but there is also a non-SB-
+		compatible 16 bit mode. Usually it's MSS/WSS but it could also
+		be a proprietary one like MV Jazz16 or ESS ES688. OPTi
+		MAD16 chips are very common in so called "SB 16 bit cards"
+		(try with the MAD16 driver).
+
+	======================================================================
+	"Supposed to be SB compatible" cards.
+		Forget the SB compatibility and check for other alternatives
+		first. The only cards that work with the SB driver in
+		Linux have been made by Creative Technology (there is at least
+		one chip on the card with "CREATIVE" printed on it). The
+		only other SB compatible chips are ESS and Jazz16 chips
+		(maybe ALSxxx chips too but they probably don't work).
+		Most other "16 bit SB compatible" cards such as "OPTi/MAD16" or
+		"Crystal" are _NOT_ SB compatible in Linux.
+
+		Practically all sound cards have some kind of SB emulation mode
+		in addition to their native (16 bit) mode. In most cases this
+		(8 bit only) SB compatible mode doesn't work with Linux. If
+		you get it working it may cause problems with games and
+		applications which require 16 bit audio. Some 16 bit only
+		applications don't check if the card actually supports 16 bits.
+		They just dump 16 bit data to a 8 bit card which produces just
+		noise.
+
+		In most cases the 16 bit native mode is supported by Linux.
+		Use the SB mode with "clones" only if you don't find anything
+		better from the rest of this doc.
+	======================================================================
+
+Gravis Ultrasound (GUS)
+	GUS
+	GUS + the 16 bit option
+	GUS MAX
+	GUS ACE (No MIDI port and audio recording)
+	GUS PnP (with RAM)
+
+MPU-401	and compatibles
+	The driver works both with the full (intelligent mode) MPU-401
+	cards (such as MPU IPC-T and MQX-32M) and with the UART only
+	dumb MIDI ports. MPU-401 is currently the most common MIDI
+	interface. Most sound cards are compatible with it. However,
+	don't enable MPU401 mode blindly. Many cards with native support
+	in the driver have their own MPU401 driver. Enabling the standard one
+	will cause a conflict with these cards. So check if your card is
+	in the list of supported cards before enabling MPU401.
+
+Windows Sound System (MSS/WSS)
+	Even when Microsoft has discontinued their own Sound System card 
+	they managed to make it a standard. MSS compatible cards are based on 
+	a codec chip which is easily available from at least two manufacturers
+	(AD1848 by Analog Devices and CS4231/CS4248 by Crystal Semiconductor).
+	Currently most sound cards are based on one of the MSS compatible codec
+	chips. The CS4231 is used in the high quality cards such as GUS MAX,
+	MediaTrix AudioTrix Pro and TB Tropez (GUS MAX is not MSS compatible). 
+
+	Having a AD1848, CS4248 or CS4231 codec chip on the card is a good
+	sign. Even if the card is not MSS compatible, it could be easy to write
+	support for it. Note also that most MSS compatible cards
+	require special boot time initialization which may not be present
+	in the driver. Also, some MSS compatible cards have native support.
+	Enabling the MSS support with these cards is likely to
+	cause a conflict. So check if your card is listed in this file before
+	enabling the MSS support.
+
+Yamaha FM synthesizers (OPL2, OPL3 (not OPL3-SA) and OPL4)
+	Most sound cards have a FM synthesizer chip. The OPL2 is a 2
+	operator chip used in the original AdLib card. Currently it's used
+	only in the cheapest (8 bit mono) cards. The OPL3 is a 4 operator 
+	FM chip which provides better sound quality and/or more available 
+	voices than the OPL2. The OPL4 is a new chip that has an OPL3 and
+	a wave table synthesizer packed onto the same chip. The driver supports
+	just the OPL3 mode directly. Most cards with an OPL4 (like
+	SM Wave and AudioTrix Pro) support the OPL4 mode using MPU401
+	emulation. Writing a native OPL4 support is difficult
+	since Yamaha doesn't give information about their sample ROM chip.
+
+	Enable the generic OPL2/OPL3 FM synthesizer support if your
+	card has a FM chip made by Yamaha. Don't enable it if your card
+	has a software (TRS) based FM emulator.
+
+	----------------------------------------------------------------
+	NOTE! OPL3-SA is different chip than the ordinary OPL3. In addition
+	to the FM synth this chip has also digital audio (WSS) and
+	MIDI (MPU401) capabilities. Support for OPL3-SA is described below.
+	----------------------------------------------------------------
+
+Yamaha OPL3-SA1
+
+	Yamaha OPL3-SA1 (YMF701) is an audio controller chip used on some
+	(Intel) motherboards and on cheap sound cards. It should not be
+	confused with the original OPL3 chip (YMF278) which is entirely
+        different chip. OPL3-SA1 has support for MSS, MPU401 and SB Pro
+	(not used in OSS/Free) in addition to the OPL3 FM synth.
+
+	There are also chips called OPL3-SA2, OPL3-SA3, ..., OPL3SA-N. They
+	are PnP chips and will not work with the OPL3-SA1 driver. You should 
+	use the standard MSS, MPU401 and OPL3 options with these chips and to
+	activate the card using isapnptools.
+
+4Front Technologies SoftOSS
+
+	SoftOSS is a software based wave table emulation which works with
+	any 16 bit stereo sound card. Due to its nature a fast CPU is
+	required (P133 is minimum). Although SoftOSS does _not_ use MMX
+	instructions it has proven out that recent processors (which appear
+	to have MMX) perform significantly better with SoftOSS than earlier
+	ones. For example a P166MMX beats a PPro200. SoftOSS should not be used
+	on 486 or 386 machines.
+
+	The amount of CPU load caused by SoftOSS can be controlled by
+	selecting the CONFIG_SOFTOSS_RATE and CONFIG_SOFTOSS_VOICES
+	parameters properly (they will be prompted by make config). It's
+	recommended to set CONFIG_SOFTOSS_VOICES to 32. If you have a
+	P166MMX or faster (PPro200 is not faster) you can set
+	CONFIG_SOFTOSS_RATE to 44100 (kHz). However with slower systems it
+	recommended to use sampling rates around 22050 or even 16000 kHz.
+	Selecting too high values for these parameters may hang your
+	system when playing MIDI files with hight degree of polyphony
+	(number of concurrently playing notes). It's also possible to
+	decrease CONFIG_SOFTOSS_VOICES. This makes it possible to use
+	higher sampling rates. However using fewer voices decreases
+	playback quality more than decreasing the sampling rate.
+
+	SoftOSS keeps the samples loaded on the system's RAM so much RAM is
+	required. SoftOSS should never be used on machines with less than 16 MB
+	of RAM since this is potentially dangerous (you may accidentally run out
+	of memory which probably crashes the machine). 
+
+	SoftOSS implements the wave table API originally designed for GUS. For
+	this reason all applications designed for GUS should work (at least
+	after minor modifications). For example gmod/xgmod and playmidi -g are
+	known to work.
+
+	To work SoftOSS will require GUS compatible
+	patch files to be installed on the system (in /dos/ultrasnd/midi). You
+	can use the public domain MIDIA patchset available from several ftp
+	sites.
+
+        *********************************************************************
+	IMPORTANT NOTICE! The original patch set distributed with the Gravis 
+	Ultrasound card is not in public domain (even though it's available from
+	some FTP sites). You should contact Voice Crystal (www.voicecrystal.com)
+	if you like to use these patches with SoftOSS included in OSS/Free.
+        *********************************************************************
+
+PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC)
+	Analog Devices and Echo Speech have together defined a sound card
+	architecture based on the above chips. The DSP chip is used
+	for emulation of SB Pro, FM and General MIDI/MT32.
+
+	There are several cards based on this architecture. The most known
+	ones are Orchid SW32 and Cardinal DSP16. 
+
+	The driver supports downloading DSP algorithms to these cards.
+
+	NOTE! You will have to use the "old" config script when configuring
+	PSS cards.
+
+MediaTrix AudioTrix Pro
+	The ATP card is built around a CS4231 codec and an OPL4 synthesizer
+	chips. The OPL4 mode is supported by a microcontroller running a
+	General MIDI emulator. There is also a SB 1.5 compatible playback mode.
+
+Ensoniq SoundScape and compatibles
+	Ensoniq has designed a sound card architecture based on the
+	OTTO synthesizer chip used in their professional MIDI synthesizers.
+	Several companies (including Ensoniq, Reveal and Spea) are selling
+	cards based on this architecture.
+
+	NOTE! The SoundScape PnP is not supported by OSS/Free. Ensoniq VIVO and
+	VIVO90 cards are not compatible with Soundscapes so the Soundscape
+        driver will not work with them. You may want to use OSS/Linux with these
+        cards.
+
+OPTi MAD16 and Mozart based cards
+	The Mozart (OAK OTI-601), MAD16 (OPTi 82C928), MAD16 Pro (OPTi 82C929),
+	OPTi 82C924/82C925 (in _non_ PnP mode) and OPTi 82C930 interface
+	chips are used in many different sound cards, including some
+	cards by Reveal miro and Turtle Beach (Tropez). The purpose of these
+	chips is to connect other audio components to the PC bus. The
+	interface chip performs address decoding for the other chips.
+	NOTE! Tropez Plus is not MAD16 but CS4232 based.
+	NOTE! MAD16 PnP cards (82C924, 82C925, 82C931) are not MAD16 compatible
+	in the PnP mode. You will have to use them in MSS mode after having
+	initialized them using isapnptools or DOS. 82C931 probably requires
+	initialization using DOS/Windows (running isapnptools is not enough).
+	It's possible to use 82C931 with OSS/Free by jumpering it to non-PnP
+	mode (provided that the card has a jumper for this). In non-PnP mode
+	82C931 is compatible with 82C930 and should work with the MAD16 driver
+	(without need to use isapnptools or DOS to initialize it). All OPTi
+	chips are supported by OSS/Linux (both in PnP and non-PnP modes).
+
+Audio Excel DSP16 
+	Support for this card was written by Riccardo Faccetti
+	(riccardo@cdc8g5.cdc.polimi.it). The AEDSP16 driver included in
+	the lowlevel/ directory. To use it you should enable the
+	"Additional low level drivers" option.
+
+Crystal CS4232 and CS4236 based cards such as AcerMagic S23, TB Tropez _Plus_ and 
+	many PC motherboards (Compaq, HP, Intel, ...)
+	CS4232 is a PnP multimedia chip which contains a CS3231A codec,
+	SB and MPU401 emulations. There is support for OPL3 too.
+	Unfortunately the MPU401 mode doesn't work (I don't know how to
+	initialize it). CS4236 is an enhanced (compatible) version of CS4232.
+	NOTE! Don't ever try to use isapnptools with CS4232 since this will just
+	freeze your machine (due to chip bugs). If you have problems in getting
+	CS4232 working you could try initializing it with DOS (CS4232C.EXE) and
+	then booting Linux using loadlin. CS4232C.EXE loads a secret firmware
+	patch which is not documented by Crystal.
+
+Turtle Beach Maui and Tropez "classic"
+	This driver version supports sample, patch and program loading commands
+	described in the Maui/Tropez User's manual. 
+	There is now full initialization support too. The audio side of
+	the Tropez is based on the MAD16 chip (see above).
+	NOTE! Tropez Plus is different card than Tropez "classic" and will not
+	work fully in Linux. You can get audio features working by configuring
+	the card as a CS4232 based card (above).
+
+
+Jumpers and software configuration
+==================================
+
+Some of the earliest sound cards were jumper configurable. You have to
+configure the driver use I/O, IRQ and DMA settings
+that match the jumpers. Just few 8 bit cards are fully jumper 
+configurable (SB 1.x/2.x, SB Pro and clones).
+Some cards made by Aztech have an EEPROM which contains the 
+config info. These cards behave much like hardware jumpered cards.
+
+Most cards have jumper for the base I/O address but other parameters
+are software configurable. Sometimes there are few other jumpers too.
+
+Latest cards are fully software configurable or they are PnP ISA
+compatible. There are no jumpers on the board.
+
+The driver handles software configurable cards automatically. Just configure
+the driver to use I/O, IRQ and DMA settings which are known to work.
+You could usually use the same values than with DOS and/or Windows.
+Using different settings is possible but not recommended since it may cause
+some trouble (for example when warm booting from an OS to another or
+when installing new hardware to the machine).
+
+Sound driver sets the soft configurable parameters of the card automatically
+during boot. Usually you don't need to run any extra initialization
+programs when booting Linux but there are some exceptions. See the
+card-specific instructions below for more info.
+
+The drawback of software configuration is that the driver needs to know
+how the card must be initialized. It cannot initialize unknown cards
+even if they are otherwise compatible with some other cards (like SB,
+MPU401 or Windows Sound System).
+
+
+What if your card was not listed above?
+=======================================
+
+The first thing to do is to look at the major IC chips on the card.
+Many of the latest sound cards are based on some standard chips. If you
+are lucky, all of them could be supported by the driver. The most common ones
+are the OPTi MAD16, Mozart, SoundScape (Ensoniq) and the PSS architectures
+listed above. Also look at the end of this file for list of unsupported
+cards and the ones which could be supported later.
+
+The last resort is to send _exact_ name and model information of the card
+to me together with a list of the major IC chips (manufactured, model) to 
+me. I could then try to check if your card looks like something familiar.
+
+There are many more cards in the world than listed above. The first thing to
+do with these cards is to check if they emulate some other card or interface
+such as SB, MSS and/or MPU401. In this case there is a chance to get the
+card to work by booting DOS before starting Linux (boot DOS, hit ctrl-alt-del
+and boot Linux without hard resetting the machine). In this method the
+DOS based driver initializes the hardware to use known I/O, IRQ and DMA
+settings. If sound driver is configured to use the same settings, everything
+should work OK.
+
+
+Configuring sound driver (with Linux)
+=====================================
+
+The sound driver is currently distributed as part of the Linux kernel. The 
+files are in /usr/src/linux/drivers/sound/. 
+
+****************************************************************************
+*	ALWAYS USE THE SOUND DRIVER VERSION WHICH IS DISTRIBUTED WITH	   *
+*	THE KERNEL SOURCE PACKAGE YOU ARE USING. SOME ALPHA AND BETA TEST  *
+*	VERSIONS CAN BE INSTALLED FROM A SEPARATELY DISTRIBUTED PACKAGE	   *
+*	BUT CHECK THAT THE PACKAGE IS NOT MUCH OLDER (OR NEWER) THAN THE   *
+*	KERNEL YOU ARE USING. IT'S POSSIBLE THAT THE KERNEL/DRIVER 	   *
+*	INTERFACE CHANGES BETWEEN KERNEL RELEASES WHICH MAY CAUSE SOME	   *
+*	INCOMPATIBILITY PROBLEMS.					   *
+*									   *
+*	IN CASE YOU INSTALL A SEPARATELY DISTRIBUTED SOUND DRIVER VERSION, *
+*	BE SURE TO REMOVE OR RENAME THE OLD SOUND DRIVER DIRECTORY BEFORE  *
+*	INSTALLING THE NEW ONE. LEAVING OLD FILES TO THE SOUND DRIVER	   *
+*	DIRECTORY _WILL_ CAUSE PROBLEMS WHEN THE DRIVER IS USED OR	   *
+*	COMPILED.							   *
+****************************************************************************
+
+To configure the driver, run "make config" in the kernel source directory
+(/usr/src/linux). Answer "y" or "m" to the question about Sound card support
+(after the questions about mouse, CD-ROM, ftape, etc. support).  Questions
+about options for sound will then be asked.
+
+After configuring the kernel and sound driver and compile the kernel 
+following instructions in the kernel README.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the first question (PAS16) if you don't really have a PAS16. Don't enable
+more cards than you really need since they just consume memory. Also
+some drivers (like MPU401) may conflict with your SCSI controller and
+prevent kernel from booting. If you card was in the list of supported
+cards (above), please look at the card specific config instructions
+(later in this file) before starting to configure. Some cards must be
+configured in way which is not obvious.
+
+So here is the beginning of the config dialog. Answer 'y' or 'n' to these
+questions. The default answer is shown so that (y/n) means 'y' by default and
+(n/y) means 'n'. To use the default value, just hit ENTER. But be careful
+since using the default _doesn't_ guarantee anything.
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+  "ProAudioSpectrum 16 support",
+	- Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+	  Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+	  you read the above list correctly). Don't answer 'y' if you
+	  have some other card made by Media Vision or Logitech since they
+	  are not PAS16 compatible.
+	  NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+	  if you want to use the SB emulation of PAS16. It's also possible to
+   	  the emulation if you want to use a true SB card together with PAS16
+	  (there is another question about this that is asked later).
+  "Sound Blaster support",
+	- Answer 'y' if you have an original SB card made by Creative Labs
+	  or a full 100% hardware compatible clone (like Thunderboard or
+	  SM Games). If your card was in the list of supported cards (above),
+	  please look at the card specific instructions later in this file
+	  before answering this question. For an unknown card you may answer 
+	  'y' if the card claims to be SB compatible.
+	 Enable this option also with PAS16 (changed since v3.5-beta9).
+
+	 Don't enable SB if you have a MAD16 or Mozart compatible card.
+
+  "Generic OPL2/OPL3 FM synthesizer support",
+	- Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+	  Answering 'y' is usually a safe and recommended choice. However some
+	  cards may have software (TSR) FM emulation. Enabling FM support
+	  with these cards may cause trouble. However I don't currently know
+	  such cards.
+  "Gravis Ultrasound support",
+	- Answer 'y' if you have GUS or GUS MAX. Answer 'n' if you don't
+	  have GUS since the GUS driver consumes much memory.
+	  Currently I don't have experiences with the GUS ACE so I don't
+	  know what to answer with it.
+  "MPU-401 support (NOT for SB16)",
+	- Be careful with this question. The MPU401 interface is supported
+	  by almost any sound card today. However some natively supported cards
+	  have their own driver for MPU401. Enabling the MPU401 option with
+	  these cards will cause a conflict. Also enabling MPU401 on a system
+	  that doesn't really have a MPU401 could cause some trouble. If your
+	  card was in the list of supported cards (above), please look at
+	  the card specific instructions later in this file.
+
+	  In MOST cases this MPU401 driver should only be used with "true"
+	  MIDI-only MPU401 professional cards. In most other cases there
+	  is another way to get the MPU401 compatible interface of a
+	  sound card to work.
+	  Support for the MPU401 compatible MIDI port of SB16, ESS1688
+	  and MV Jazz16 cards is included in the SB driver. Use it instead
+	  of this separate MPU401 driver with these cards. As well 
+	  Soundscape, PSS and Maui drivers include their own MPU401
+	  options.
+
+	  It's safe to answer 'y' if you have a true MPU401 MIDI interface
+	  card. 
+  "6850 UART Midi support",
+	- It's safe to answer 'n' to this question in all cases. The 6850
+	  UART interface is so rarely used.
+  "PSS (ECHO-ADI2111) support",
+	- Answer 'y' only if you have Orchid SW32, Cardinal DSP16 or some
+	  other card based on the PSS chipset (AD1848 codec + ADSP-2115
+	  DSP chip + Echo ESC614 ASIC CHIP).
+  "16 bit sampling option of GUS (_NOT_ GUS MAX)",
+	- Answer 'y' if you have installed the 16 bit sampling daughtercard
+	  to your GUS. Answer 'n' if you have GUS MAX. Enabling this option
+	  disables GUS MAX support.
+  "GUS MAX support",
+	- Answer 'y' only if you have a GUS MAX.
+  "Microsoft Sound System support",
+	- Again think carefully before answering 'y' to this question. It's
+	  safe to answer 'y' in case you have the original Windows Sound 
+	  System card made by Microsoft or Aztech SG 16 Pro (or NX16 Pro).
+	  Also you may answer 'y' in case your card was not listed earlier
+	  in this file. For cards having native support in the driver, consult
+	  the card specific instructions later in this file. Some drivers
+	  have their own MSS support and enabling this option will cause a
+	  conflict. 
+	  Note! The MSS driver permits configuring two DMA channels. This is a
+	  "nonstandard" feature and works only with very few cards (if any).
+	  In most cases the second DMA channel should be disabled or set to
+	  the same channel than the first one. Trying to configure two separate
+	  channels with cards that don't support this feature will prevent
+	  audio (at least recording) from working.
+  "Ensoniq Soundscape support",
+	- Answer 'y' if you have a sound card based on the Ensoniq SoundScape
+	  chipset. Such cards are being manufactured at least by Ensoniq,
+	  Spea and Reveal (note that Reveal makes other cards also).  The oldest
+	  cards made by Spea don't work properly with Linux. 
+	  Soundscape PnP as well as Ensoniq VIVO work only with the commercial
+	  OSS/Linux version.
+  "MediaTrix AudioTrix Pro support",
+	- Answer 'y' if you have the AudioTrix Pro.
+  "Support for MAD16 and/or Mozart based cards",
+	- Answer y if your card has a Mozart (OAK OTI-601) or MAD16
+	  (OPTi 82C928, 82C929, 82C924/82C925 or 82C930) audio interface chip. 
+	  These chips are
+	  currently quite common so it's possible that many no-name cards
+	  have one of them. In addition the MAD16 chip is used in some
+	  cards made by known manufacturers such as Turtle Beach (Tropez),
+	  Reveal (some models) and Diamond (some recent models).
+	  Note OPTi 82C924 and 82C925 are MAD16 compatible only in non PnP
+	  mode (jumper selectable on many cards).
+  "Support for TB Maui"
+	- This enables TB Maui specific initialization. Works with TB Maui
+	and TB Tropez (may not work with Tropez Plus).
+
+
+Then the configuration program asks some y/n questions about the higher
+level services. It's recommended to answer 'y' to each of these questions.
+Answer 'n' only if you know you will not need the option.
+
+  "MIDI interface support",
+	- Answering 'n' disables /dev/midi## devices and access to any
+	  MIDI ports using /dev/sequencer and /dev/music. This option
+	  also affects any MPU401 and/or General MIDI compatible devices.
+  "FM synthesizer (YM3812/OPL-3) support",
+	- Answer 'y' here.
+  "/dev/sequencer support",
+	- Answering 'n' disables /dev/sequencer and /dev/music.
+
+Entering the I/O, IRQ and DMA config parameters
+-----------------------------------------------
+
+After the above questions the configuration program prompts for the
+card specific configuration information. Usually just a set of
+I/O address, IRQ and DMA numbers are asked. With some cards the program
+asks for some files to be used during initialization of the card. For example
+many cards have a DSP chip or microprocessor which must be initialized by
+downloading a program (microcode) file to the card.
+
+Instructions for answering these questions are given in the next section.
+
+
+Card specific information
+=========================
+
+This section gives additional instructions about configuring some cards.
+Please refer manual of your card for valid I/O, IRQ and DMA numbers. Using
+the same settings with DOS/Windows and Linux is recommended. Using
+different values could cause some problems when switching between
+different operating systems.
+
+Sound Blasters (the original ones by Creative)
+---------------------------------------------
+
+NOTE! Check if you have a PnP Sound Blaster (cards sold after summer 1995
+      are almost certainly PnP ones). With PnP cards you should use isapnptools
+      to activate them (see above).
+
+It's possible to configure these cards to use different I/O, IRQ and
+DMA settings. Since the possible/default settings have changed between various
+models, you have to consult manual of your card for the proper ones. It's
+a good idea to use the same values than with DOS/Windows. With SB and SB Pro
+it's the only choice. SB16 has software selectable IRQ and DMA channels but
+using different values with DOS and Linux is likely to cause troubles. The
+DOS driver is not able to reset the card properly after warm boot from Linux
+if Linux has used different IRQ or DMA values.
+
+The original (steam) Sound Blaster (versions 1.x and 2.x) use always
+DMA1. There is no way to change it.
+
+The SB16 needs two DMA channels. A 8 bit one (1 or 3) is required for
+8 bit operation and a 16 bit one (5, 6 or 7) for the 16 bit mode. In theory
+it's possible to use just one (8 bit) DMA channel by answering the 8 bit
+one when the configuration program asks for the 16 bit one. This may work
+in some systems but is likely to cause terrible noise on some other systems.
+
+It's possible to use two SB16/32/64 at the same time. To do this you should
+first configure OSS/Free for one card. Then edit local.h manually and define
+SB2_BASE, SB2_IRQ, SB2_DMA and SB2_DMA2 for the second one. You can't get
+the OPL3, MIDI and EMU8000 devices of the second card to work. If you are
+going to use two PnP Sound Blasters, ensure that they are of different model
+and have different PnP IDs. There is no way to get two cards with the same
+card ID and serial number to work. The easiest way to check this is trying 
+if isapnptools can see both cards or just one.
+
+NOTE!	Don't enable the SM Games option (asked by the configuration program)
+	if you are not 101% sure that your card is a Logitech Soundman Games
+	(not a SM Wave or SM16).
+
+SB Clones
+---------
+
+First of all: There are no SB16 clones. There are SB Pro clones with a
+16 bit mode which is not SB16 compatible. The most likely alternative is that
+the 16 bit mode means MSS/WSS.
+
+There are just a few fully 100% hardware SB or SB Pro compatible cards.
+I know just Thunderboard and SM Games. Other cards require some kind of
+hardware initialization before they become SB compatible. Check if your card
+was listed in the beginning of this file. In this case you should follow
+instructions for your card later in this file.
+
+For other not fully SB clones you may try initialization using DOS in
+the following way:
+
+	- Boot DOS so that the card specific driver gets run.
+	- Hit ctrl-alt-del (or use loadlin) to boot Linux. Don't
+	  switch off power or press the reset button.
+	- If you use the same I/O, IRQ and DMA settings in Linux, the
+	  card should work.
+
+If your card is both SB and MSS compatible, I recommend using the MSS mode.
+Most cards of this kind are not able to work in the SB and the MSS mode 
+simultaneously. Using the MSS mode provides 16 bit recording and playback.
+
+ProAudioSpectrum 16 and compatibles
+-----------------------------------
+
+PAS16 has a SB emulation chip which can be used together with the native
+(16 bit) mode of the card. To enable this emulation you should configure 
+the driver to have SB support too (this has been changed since version
+3.5-beta9 of this driver).
+
+With current driver versions it's also possible to use PAS16 together with 
+another SB compatible card. In this case you should configure SB support
+for the other card and to disable the SB emulation of PAS16 (there is a
+separate questions about this).
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+Gravis Ultrasound
+-----------------
+
+There are many different revisions of the Ultrasound card (GUS). The
+earliest ones (pre 3.7) don't have a hardware mixer. With these cards
+the driver uses a software emulation for synth and pcm playbacks. It's
+also possible to switch some of the inputs (line in, mic) off by setting
+mixer volume of the channel level below 10%. For recording you have
+to select the channel as a recording source and to use volume above 10%.
+
+GUS 3.7 has a hardware mixer.
+
+GUS MAX and the 16 bit sampling daughtercard have a CS4231 codec chip which 
+also contains a mixer.
+
+Configuring GUS is simple. Just enable the GUS support and GUS MAX or
+the 16 bit daughtercard if you have them. Note that enabling the daughter
+card disables GUS MAX driver.
+
+NOTE for owners of the 16 bit daughtercard: By default the daughtercard
+uses /dev/dsp (and /dev/audio). Command "ln -sf /dev/dsp1 /dev/dsp"
+selects the daughter card as the default device.
+
+With just the standard GUS enabled the configuration program prompts
+for the I/O, IRQ and DMA numbers for the card. Use the same values than
+with DOS.
+
+With the daughter card option enabled you will be prompted for the I/O,
+IRQ and DMA numbers for the daughter card. You have to use different I/O
+and DMA values than for the standard GUS. The daughter card permits
+simultaneous recording and playback. Use /dev/dsp (the daughtercard) for
+recording and /dev/dsp1 (GUS GF1) for playback.
+
+GUS MAX uses the same I/O address and IRQ settings than the original GUS
+(GUS MAX = GUS + a CS4231 codec). In addition an extra DMA channel may be used.
+Using two DMA channels permits simultaneous playback using two devices
+(dev/dsp0 and /dev/dsp1). The second DMA channel is required for
+full duplex audio.
+To enable the second DMA channels, give a valid DMA channel when the config
+program asks for the GUS MAX DMA (entering -1 disables the second DMA).
+Using 16 bit DMA channels (5,6 or 7) is recommended.
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
+
+Microphone input of GUS MAX is connected to mixer in little bit nonstandard
+way. There is actually two microphone volume controls. Normal "mic" controls
+only recording level. Mixer control "speaker" is used to control volume of
+microphone signal connected directly to line/speaker out. So just decrease
+volume of "speaker" if you have problems with microphone feedback.
+
+GUS ACE works too but any attempt to record or to use the MIDI port
+will fail.
+
+GUS PnP (with RAM) is partially supported but it needs to be initialized using
+DOS or isapnptools before starting the driver.
+
+MPU401 and Windows Sound System
+-------------------------------
+
+Again. Don't enable these options in case your card is listed
+somewhere else in this file.
+
+Configuring these cards is obvious (or it should be). With MSS
+you should probably enable the OPL3 synth also since
+most MSS compatible cards have it. However check that this is true
+before enabling OPL3.
+
+Sound driver supports more than one MPU401 compatible cards at the same time
+but the config program asks config info for just the first of them.
+Adding the second or third MPU interfaces must be done manually by
+editing sound/local.h (after running the config program). Add defines for
+MPU2_BASE & MPU2_IRQ (and MPU3_BASE & MPU3_IRQ) to the file.
+
+CAUTION!
+
+The default I/O base of Adaptec AHA-1542 SCSI controller is 0x330 which
+is also the default of the MPU401 driver. Don't configure the sound driver to
+use 0x330 as the MPU401 base if you have a AHA1542. The kernel will not boot
+if you make this mistake.
+
+PSS
+---
+
+Even the PSS cards are compatible with SB, MSS and MPU401, you must not
+enable these options when configuring the driver. The configuration
+program handles these options itself. (You may use the SB, MPU and MSS options
+together with PSS if you have another card on the system).
+
+The PSS driver enables MSS and MPU401 modes of the card. SB is not enabled 
+since it doesn't work concurrently with MSS. The driver loads also a
+DSP algorithm which is used to for the general MIDI emulation. The
+algorithm file (.ld) is read by the config program and written to a
+file included when the pss.c is compiled. For this reason the config
+program asks if you want to download the file. Use the genmidi.ld file
+distributed with the DOS/Windows drivers of the card (don't use the mt32.ld).
+With some cards the file is called 'synth.ld'. You must have access to
+the file when configuring the driver. The easiest way is to mount the DOS
+partition containing the file with Linux. 
+
+It's possible to load your own DSP algorithms and run them with the card.
+Look at the directory pss_test of snd-util-3.0.tar.gz for more info.
+
+AudioTrix Pro
+-------------
+
+You have to enable the OPL3 and SB (not SB Pro or SB16) drivers in addition
+to the native AudioTrix driver. Don't enable MSS or MPU drivers.
+
+Configuring ATP is little bit tricky since it uses so many I/O, IRQ and
+DMA numbers. Using the same values than with DOS/Win is a good idea. Don't
+attempt to use the same IRQ or DMA channels twice.
+
+The SB mode of ATP is implemented so the ATP driver just enables SB
+in the proper address. The SB driver handles the rest. You have to configure
+both the SB driver and the SB mode of ATP to use the same IRQ, DMA and I/O
+settings.
+
+Also the ATP has a microcontroller for the General MIDI emulation (OPL4).
+For this reason the driver asks for the name of a file containing the
+microcode (TRXPRO.HEX). This file is usually located in the directory
+where the DOS drivers were installed. You must have access to this file
+when configuring the driver. 
+
+If you have the effects daughtercard, it must be initialized by running
+the setfx program of snd-util-3.0.tar.gz package. This step is not required
+when using the (future) binary distribution version of the driver.
+
+Ensoniq SoundScape
+------------------
+
+NOTE!	The new PnP SoundScape is not supported yet. Soundscape compatible
+	cards made by Reveal don't work with Linux. They use older revision
+	of the Soundscape chipset which is not fully compatible with
+	newer cards made by Ensoniq.
+
+The SoundScape driver handles initialization of MSS and MPU supports
+itself so you don't need to enable other drivers than SoundScape
+(enable also the /dev/dsp, /dev/sequencer and MIDI supports).
+
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+!!!!!                                                                   !!!!
+!!!!! NOTE! Before version 3.5-beta6 there WERE two sets of audio 	!!!!
+!!!!!       device files (/dev/dsp0 and /dev/dsp1). The first one WAS   !!!!
+!!!!!	    used only for card initialization and the second for audio  !!!!
+!!!!!	    purposes. It WAS required to change /dev/dsp (a symlink) to !!!!
+!!!!!	    point to /dev/dsp1.						!!!!
+!!!!!                                                                   !!!!
+!!!!!	    This is not required with OSS versions 3.5-beta6 and later	!!!!
+!!!!!	    since there is now just one audio device file. Please 	!!!!
+!!!!!	    change /dev/dsp to point back to /dev/dsp0 if you are	!!!!
+!!!!!	    upgrading from an earlier driver version using		!!!!
+!!!!!	    (cd /dev;rm dsp;ln -s dsp0 dsp).				!!!!
+!!!!!                                                                   !!!!
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+
+The configuration program asks one DMA channel and two interrupts. One IRQ
+and one DMA is used by the MSS codec. The second IRQ is required for the
+MPU401 mode (you have to use different IRQs for both purposes).
+There were earlier two DMA channels for SoundScape but the current driver
+version requires just one.
+
+The SoundScape card has a Motorola microcontroller which must initialized
+_after_ boot (the driver doesn't initialize it during boot).
+The initialization is done by running the 'ssinit' program which is
+distributed in the snd-util-3.0.tar.gz package. You have to edit two
+defines in the ssinit.c and then compile the program. You may run ssinit 
+manually (after each boot) or add it to /etc/rc.d/rc.local.
+
+The ssinit program needs the microcode file that comes with the DOS/Windows
+driver of the card. You will need to use version 1.30.00 or later
+of the microcode file (sndscape.co0 or sndscape.co1 depending on
+your card model). THE OLD sndscape.cod WILL NOT WORK. IT WILL HANG YOUR
+MACHINE. The only way to get the new microcode file is to download
+and install the DOS/Windows driver from ftp://ftp.ensoniq.com/pub.
+
+Then you have to select the proper microcode file to use: soundscape.co0
+is the right one for most cards and sndscape.co1 is for few (older) cards
+made by Reveal and/or Spea. The driver has capability to detect the card
+version during boot. Look at the boot log messages in /var/adm/messages
+and locate the sound driver initialization message for the SoundScape
+card. If the driver displays string , you have
+an old card and you will need to use sndscape.co1. For other cards use
+soundscape.co0. New Soundscape revisions such as Elite and PnP use
+code files with higher numbers (.co2, .co3, etc.).
+
+NOTE!	Ensoniq Soundscape VIVO is not compatible with other Soundscape cards.
+	Currently it's possible to use it in Linux only with OSS/Linux
+	drivers.
+
+Check /var/adm/messages after running ssinit. The driver prints
+the board version after downloading the microcode file. That version
+number must match the number in the name of the microcode file (extension).
+
+Running ssinit with a wrong version of the sndscape.co? file is not
+dangerous as long as you don't try to use a file called sndscape.cod.
+If you have initialized the card using a wrong microcode file (sounds
+are terrible), just modify ssinit.c to use another microcode file and try
+again. It's possible to use an earlier version of sndscape.co[01] but it
+may sound weird.
+
+MAD16 (Pro) and Mozart
+----------------------
+
+You need to enable just the MAD16 /Mozart support when configuring
+the driver. _Don't_ enable SB, MPU401 or MSS. However you will need the
+/dev/audio, /dev/sequencer and MIDI supports.
+
+Mozart and OPTi 82C928 (the original MAD16) chips don't support
+MPU401 mode so enter just 0 when the configuration program asks the
+MPU/MIDI I/O base. The MAD16 Pro (OPTi 82C929) and 82C930 chips have MPU401
+mode.
+
+TB Tropez is based on the 82C929 chip. It has two MIDI ports.
+The one connected to the MAD16 chip is the second one (there is a second
+MIDI connector/pins somewhere??). If you have not connected the second MIDI
+port, just disable the MIDI port of MAD16. The 'Maui' compatible synth of
+Tropez is jumper configurable and not connected to the MAD16 chip (the
+Maui driver can be used with it).
+
+Some MAD16 based cards may cause feedback, whistle or terrible noise if the
+line3 mixer channel is turned too high. This happens at least with Shuttle
+Sound System. Current driver versions set volume of line3 low enough so
+this should not be a problem.
+
+If you have a MAD16 card which have an OPL4 (FM + Wave table) synthesizer
+chip (_not_ an OPL3), you have to append a line containing #define MAD16_OPL4
+to the file linux/drivers/sound/local.h (after running make config).
+
+MAD16 cards having a CS4231 codec support full duplex mode. This mode
+can be enabled by configuring the card to use two DMA channels. Possible
+DMA channel pairs are: 0&1, 1&0 and 3&0.
+
+NOTE! Cards having an OPTi 82C924/82C925 chip work with OSS/Free only in
+non-PnP mode (usually jumper selectable). The PnP mode is supported only
+by OSS/Linux.
+
+MV Jazz (ProSonic)
+------------------
+
+The Jazz16 driver is just a hack made to the SB Pro driver. However it works
+fairly well. You have to enable SB, SB Pro (_not_ SB16) and MPU401 supports
+when configuring the driver. The configuration program asks later if you
+want support for MV Jazz16 based cards (after asking SB base address). Answer
+'y' here and the driver asks the second (16 bit) DMA channel.
+
+The Jazz16 driver uses the MPU401 driver in a way which will cause
+problems if you have another MPU401 compatible card. In this case you must
+give address of the Jazz16 based MPU401 interface when the config
+program prompts for the MPU401 information. Then look at the MPU401
+specific section for instructions about configuring more than one MPU401 cards.
+
+Logitech Soundman Wave
+----------------------
+
+Read the above MV Jazz specific instructions first.
+
+The Logitech SoundMan Wave (don't confuse this with the SM16 or SM Games) is
+a MV Jazz based card which has an additional OPL4 based wave table
+synthesizer. The OPL4 chip is handled by an on board microcontroller
+which must be initialized during boot. The config program asks if
+you have a SM Wave immediately after asking the second DMA channel of jazz16.
+If you answer 'y', the config program will ask name of the file containing
+code to be loaded to the microcontroller. The file is usually called
+MIDI0001.BIN and it's located in the DOS/Windows driver directory. The file
+may also be called as TSUNAMI.BIN or something else (older cards?).
+
+The OPL4 synth will be inaccessible without loading the microcontroller code.
+
+Also remember to enable SB MPU401 support if you want to use the OPL4 mode.
+(Don't enable the 'normal' MPU401 device as with some earlier driver
+versions (pre 3.5-alpha8)).
+
+NOTE!	Don't answer 'y' when the driver asks about SM Games support
+	(the next question after the MIDI0001.BIN name). However
+	answering 'y' doesn't cause damage your computer so don't panic. 
+
+Sound Galaxies
+--------------
+
+There are many different Sound Galaxy cards made by Aztech. The 8 bit
+ones are fully SB or SB Pro compatible and there should be no problems
+with them. 
+
+The older 16 bit cards (SG Pro16, SG NX Pro16, Nova and Lyra) have
+an EEPROM chip for storing the configuration data. There is a microcontroller
+which initializes the card to match the EEPROM settings when the machine
+is powered on. These cards actually behave just like they have jumpers
+for all of the settings. Configure driver for MSS, MPU, SB/SB Pro  and OPL3 
+supports with these cards. 
+
+There are some new Sound Galaxies in the market. I have no experience with
+them so read the card's manual carefully.
+
+ESS ES1688 and ES688 'AudioDrive' based cards
+---------------------------------------------
+
+Support for these two ESS chips is embedded in the SB driver.
+Configure these cards just like SB. Enable the 'SB MPU401 MIDI port'
+if you want to use MIDI features of ES1688. ES688 doesn't have MPU mode
+so you don't need to enable it (the driver uses normal SB MIDI automatically
+with ES688).
+
+NOTE! ESS cards are not compatible with MSS/WSS so don't worry if MSS support
+of OSS doesn't work with it.
+
+There are some ES1688/688 based sound cards and (particularly) motherboards
+which use software configurable I/O port relocation feature of the chip.
+This ESS proprietary feature is supported only by OSS/Linux.
+
+There are ES1688 based cards which use different interrupt pin assignment than
+recommended by ESS (5, 7, 9/2 and 10). In this case all IRQs don't work.
+At least a card called (Pearl?) Hypersound 16 supports IRQ 15 but it doesn't
+work.
+
+ES1868 is a PnP chip which is (supposed to be) compatible with ESS1688
+probably works with OSS/Free after initialization using isapnptools.
+
+Reveal cards
+------------
+
+There are several different cards made/marketed by Reveal. Some of them
+are compatible with SoundScape and some use the MAD16 chip. You may have
+to look at the card and try to identify its origin.
+
+Diamond
+-------
+
+The oldest (Sierra Aria based) sound cards made by Diamond are not supported
+(they may work if the card is initialized using DOS). The recent (LX?)
+models are based on the MAD16 chip which is supported by the driver.
+
+Audio Excel DSP16
+-----------------
+
+Support for this card is currently not functional. A new driver for it
+should be available later this year.
+
+PCMCIA cards
+------------
+
+Sorry, can't help. Some cards may work and some don't.
+
+TI TM4000M notebooks
+--------------------
+
+These computers have a built in sound support based on the Jazz chipset.
+Look at the instructions for MV Jazz (above). It's also important to note
+that there is something wrong with the mouse port and sound at least on
+some TM models. Don't enable the "C&T 82C710 mouse port support" when
+configuring Linux. Having it enabled is likely to cause mysterious problems
+and kernel failures when sound is used.
+
+miroSOUND
+---------
+
+The miroSOUND PCM1-pro, PCM12 and PCM20 radio has been used
+successfully. These cards are based on the MAD16, OPL4, and CS4231A chips
+and everything said in the section about MAD16 cards applies here,
+too. The only major difference between the PCMxx and other MAD16 cards
+is that instead of the mixer in the CS4231 codec a separate mixer
+controlled by an on-board 80C32 microcontroller is used. Control of
+the mixer takes place via the ACI (miro's audio control interface)
+protocol that is implemented in a separate lowlevel driver. Make sure
+you compile this ACI driver together with the normal MAD16 support
+when you use a miroSOUND PCMxx card. The ACI mixer is controlled by
+/dev/mixer and the CS4231 mixer by /dev/mixer1 (depends on load
+time). Only in special cases you want to change something regularly on
+the CS4231 mixer.
+
+The miroSOUND PCM12 and PCM20 radio is capable of full duplex
+operation (simultaneous PCM replay and recording), which allows you to
+implement nice real-time signal processing audio effect software and
+network telephones. The ACI mixer has to be switched into the "solo"
+mode for duplex operation in order to avoid feedback caused by the
+mixer (input hears output signal). You can de-/activate this mode
+through toggleing the record button for the wave controller with an
+OSS-mixer.
+
+The PCM20 contains a radio tuner, which is also controlled by
+ACI. This radio tuner is supported by the ACI driver together with the
+miropcm20.o module. Also the 7-band equalizer is integrated
+(limited by the OSS-design). Development has started and maybe
+finished for the RDS decoder on this card, too. You will be able to
+read RadioText, the Programme Service name, Programme TYpe and
+others. Even the v4l radio module benefits from it with a refined
+strength value. See aci.[ch] and miropcm20*.[ch] for more details.
+
+The following configuration parameters have worked fine for the PCM12
+in Markus Kuhn's system, many other configurations might work, too:
+CONFIG_MAD16_BASE=0x530, CONFIG_MAD16_IRQ=11, CONFIG_MAD16_DMA=3, 
+CONFIG_MAD16_DMA2=0, CONFIG_MAD16_MPU_BASE=0x330, CONFIG_MAD16_MPU_IRQ=10,
+DSP_BUFFSIZE=65536, SELECTED_SOUND_OPTIONS=0x00281000.
+
+Bas van der Linden is using his PCM1-pro with a configuration that
+differs in: CONFIG_MAD16_IRQ=7, CONFIG_MAD16_DMA=1, CONFIG_MAD16_MPU_IRQ=9
+
+Compaq Deskpro XL
+-----------------
+
+The builtin sound hardware of Compaq Deskpro XL is now supported. 
+You need to configure the driver with MSS and OPL3 supports enabled.
+In addition you need to manually edit linux/drivers/sound/local.h and
+to add a line containing "#define DESKPROXL" if you used 
+make menuconfig/xconfig.
+
+Others?
+-------
+
+Since there are so many different sound cards, it's likely that I have 
+forgotten to mention many of them. Please inform me if you know yet another
+card which works with Linux, please inform me (or is anybody else
+willing to maintain a database of supported cards (just like in XF86)?).
+
+Cards not supported yet
+=======================
+
+Please check the version of sound driver you are using before 
+complaining that your card is not supported. It's possible you are 
+using a driver version which was released months before your card was
+introduced.
+
+First of all, there is an easy way to make most sound cards work with Linux.
+Just use the DOS based driver to initialize the card to a known state, then use
+loadlin.exe to boot Linux. If Linux is configured to use the same I/O, IRQ and
+DMA numbers as DOS, the card could work.
+(ctrl-alt-del can be used in place of loadlin.exe but it doesn't work with
+new motherboards). This method works also with all/most PnP sound cards.
+
+Don't get fooled with SB compatibility. Most cards are compatible with
+SB but that may require a TSR which is not possible with Linux. If
+the card is compatible with MSS, it's a better choice. Some cards
+don't work in the SB and MSS modes at the same time.
+
+Then there are cards which are no longer manufactured and/or which
+are relatively rarely used (such as the 8 bit ProAudioSpectrum
+models). It's extremely unlikely that such cards ever get supported.
+Adding support for a new card requires much work and increases time
+required in maintaining the driver (some changes need to be done
+to all low level drivers and be tested too, maybe with multiple
+operating systems). For this reason I have made a decision to not support
+obsolete cards. It's possible that someone else makes a separately 
+distributed driver (diffs) for the card. 
+
+Writing a driver for a new card is not possible if there are no 
+programming information available about the card. If you don't
+find your new card from this file, look from the home page 
+(http://www.opensound.com/ossfree). Then please contact
+manufacturer of the card and ask if they have (or are willing to)
+released technical details of the card. Do this before contacting me. I
+can only answer 'no' if there are no programming information available.
+
+I have made decision to not accept code based on reverse engineering
+to the driver. There are three main reasons: First I don't want to break
+relationships to sound card manufacturers. The second reason is that
+maintaining and supporting a driver without any specs will be a pain.
+The third reason is that companies have freedom to refuse selling their
+products to other than Windows users.
+
+Some companies don't give low level technical information about their
+products to public or at least their require signing a NDA. It's not
+possible to implement a freeware driver for them. However it's possible
+that support for such cards become available in the commercial version
+of this driver (see http://www.4Front-tech.com/oss.html for more info).
+
+There are some common audio chipsets that are not supported yet. For example
+Sierra Aria and IBM Mwave. It's possible that these architectures
+get some support in future but I can't make any promises. Just look
+at the home page (http://www.opensound.com/ossfree/)
+for latest info.
+
+Information about unsupported sound cards and chipsets is welcome as well
+as free copies of sound cards, SDKs and operating systems.
+
+If you have any corrections and/or comments, please contact me.
+
+Hannu Savolainen
+hannu@opensound.com
+
+home page of OSS/Free: http://www.opensound.com/ossfree
+
+home page of commercial OSS
+(Open Sound System) drivers: http://www.opensound.com/oss.html
diff --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules
new file mode 100644
index 00000000..e691d74e
--- /dev/null
+++ b/Documentation/sound/oss/README.modules
@@ -0,0 +1,106 @@
+Building a modular sound driver
+================================
+
+  The following information is current as of linux-2.1.85. Check the other
+readme files, especially README.OSS, for information not specific to
+making sound modular.
+
+  First, configure your kernel. This is an idea of what you should be
+setting in the sound section:
+
+ Sound card support 
+
+ 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support 
+
+  I have SoundBlaster. Select your card from the list.
+
+ Generic OPL2/OPL3 FM synthesizer support 
+ FM synthesizer (YM3812/OPL-3) support 
+
+  If you don't set these, you will probably find you can play .wav files
+but not .midi. As the help for them says, set them unless you know your
+card does not use one of these chips for FM support.
+
+  Once you are configured, make zlilo, modules, modules_install; reboot.
+Note that it is no longer necessary or possible to configure sound in the
+drivers/sound dir. Now one simply configures and makes one's kernel and
+modules in the usual way.
+
+ Then, add to your /etc/modprobe.conf something like:
+
+alias char-major-14-* sb
+install sb /sbin/modprobe -i sb && /sbin/modprobe adlib_card
+options sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+options adlib_card io=0x388     # FM synthesizer
+
+ Alternatively, if you have compiled in kernel level ISAPnP support:
+
+alias char-major-14 sb
+post-install sb /sbin/modprobe "-k" "adlib_card"
+options adlib_card io=0x388
+
+  The effect of this is that the sound driver and all necessary bits and
+pieces autoload on demand, assuming you use kerneld (a sound choice) and
+autoclean when not in use. Also, options for the device drivers are
+set. They will not work without them. Change as appropriate for your card.
+If you are not yet using the very cool kerneld, you will have to "modprobe
+-k sb" yourself to get things going. Eventually things may be fixed so
+that this kludgery is not necessary; for the time being, it seems to work
+well.
+
+  Replace 'sb' with the driver for your card, and give it the right
+options. To find the filename of the driver, look in
+/lib/modules//misc. Mine looks like:
+
+adlib_card.o # This is the generic OPLx driver
+opl3.o # The OPL3 driver
+sb.o # <>
+sound.o # The sound driver
+uart401.o # Used by sb, maybe other cards
+
+ Whichever card you have, try feeding it the options that would be the
+default if you were making the driver wired, not as modules. You can
+look at function referred to by module_init() for the card to see what
+args are expected.
+
+ Note that at present there is no way to configure the io, irq and other
+parameters for the modular drivers as one does for the wired drivers.. One
+needs to pass the modules the necessary parameters as arguments, either
+with /etc/modprobe.conf or with command-line args to modprobe, e.g.
+
+modprobe sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+modprobe adlib_card io=0x388
+
+ recommend using /etc/modprobe.conf.
+
+Persistent DMA Buffers:
+
+The sound modules normally allocate DMA buffers during open() and
+deallocate them during close(). Linux can often have problems allocating
+DMA buffers for ISA cards on machines with more than 16MB RAM. This is
+because ISA DMA buffers must exist below the 16MB boundary and it is quite
+possible that we can't find a large enough free block in this region after
+the machine has been running for any amount of time. The way to avoid this
+problem is to allocate the DMA buffers during module load and deallocate
+them when the module is unloaded. For this to be effective we need to load
+the sound modules right after the kernel boots, either manually or by an
+init script, and keep them around until we shut down. This is a little
+wasteful of RAM, but it guarantees that sound always works.
+
+To make the sound driver use persistent DMA buffers we need to pass the
+sound.o module a "dmabuf=1" command-line argument. This is normally done
+in /etc/modprobe.conf like so:
+
+options sound		dmabuf=1
+
+If you have 16MB or less RAM or a PCI sound card, this is wasteful and
+unnecessary. It is possible that machine with 16MB or less RAM will find
+this option useful, but if your machine is so memory-starved that it
+cannot find a 64K block free, you will be wasting even more RAM by keeping
+the sound modules loaded and the DMA buffers allocated when they are not
+needed. The proper solution is to upgrade your RAM. But you do also have
+this improper solution as well. Use it wisely.
+
+  I'm afraid I know nothing about anything but my setup, being more of a
+text-mode guy anyway. If you have options for other cards or other helpful
+hints, send them to me, Jim Bray, jb@as220.org, http://as220.org/jb.
diff --git a/Documentation/sound/oss/README.ymfsb b/Documentation/sound/oss/README.ymfsb
new file mode 100644
index 00000000..b6b77906
--- /dev/null
+++ b/Documentation/sound/oss/README.ymfsb
@@ -0,0 +1,107 @@
+Legacy audio driver for YMF7xx PCI cards.
+
+
+FIRST OF ALL
+============
+
+  This code references YAMAHA's sample codes and data sheets.
+  I respect and thank for all people they made open the information
+  about YMF7xx cards.
+
+  And this codes heavily based on Jeff Garzik 's
+  old VIA 82Cxxx driver (via82cxxx.c). I also respect him.
+
+
+DISCLIMER
+=========
+
+ This driver is currently at early ALPHA stage. It may cause serious
+ damage to your computer when used.
+ PLEASE USE IT AT YOUR OWN RISK.
+
+
+ABOUT THIS DRIVER
+=================
+
+  This code enables you to use your YMF724[A-F], YMF740[A-C], YMF744, YMF754
+ cards. When enabled, your card acts as "SoundBlaster Pro" compatible card.
+ It can only play 22.05kHz / 8bit / Stereo samples, control external MIDI
+ port.
+ If you want to use your card as recent "16-bit" card, you should use
+ Alsa or OSS/Linux driver. Of course you can write native PCI driver for
+ your cards :)
+
+
+USAGE
+=====
+
+ # modprobe ymfsb (options)
+
+
+OPTIONS FOR MODULE
+==================
+
+  io           : SB base address     (0x220, 0x240, 0x260, 0x280)
+  synth_io     : OPL3 base address   (0x388, 0x398, 0x3a0, 0x3a8)
+  dma          : DMA number          (0,1,3)
+  master_volume: AC'97 PCM out Vol   (0-100)
+  spdif_out    : SPDIF-out flag      (0:disable 1:enable)
+
+ These options will change in future...
+
+
+FREQUENCY
+=========
+
+  When playing sounds via this driver, you will hear its pitch is slightly
+ lower than original sounds. Since this driver recognizes your card acts
+ with 21.739kHz sample rates rather than 22.050kHz (I think it must be
+ hardware restriction). So many players become tone deafness.
+ To prevent this, you should express some options to your sound player
+ that specify correct sample frequency. For example, to play your MP3 file
+ correctly with mpg123, specify the frequency like following:
+
+  %  mpg123 -r 21739 foo.mp3
+
+
+SPDIF OUT
+=========
+
+  With installing modules with option 'spdif_out=1', you can enjoy your
+ sounds from SPDIF-out of your card (if it had).
+ Its Fs is fixed to 48kHz (It never means the sample frequency become 
+ up to 48kHz. All sounds via SPDIF-out also 22kHz samples). So your
+ digital-in capable components has to be able to handle 48kHz Fs.
+
+
+COPYING
+=======
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2, or (at your option)
+ any later version.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+
+TODO
+====
+ * support for multiple cards
+   (set the different SB_IO,MPU_IO,OPL_IO for each cards)
+
+ * support for OPL (dmfm) : There will be no requirements... :-<
+
+
+AUTHOR
+======
+
+ Daisuke Nagano 
+
diff --git a/Documentation/sound/oss/SoundPro b/Documentation/sound/oss/SoundPro
new file mode 100644
index 00000000..9d4db1f2
--- /dev/null
+++ b/Documentation/sound/oss/SoundPro
@@ -0,0 +1,105 @@
+Documentation for the SoundPro CMI8330 extensions in the WSS driver (ad1848.o)
+------------------------------------------------------------------------------
+
+( Be sure to read Documentation/sound/oss/CMI8330 too )
+
+Ion Badulescu, ionut@cs.columbia.edu
+February 24, 1999
+
+(derived from the OPL3-SA2 documentation by Scott Murray)
+
+The SoundPro CMI8330 (ISA) is a chip usually found on some Taiwanese
+motherboards.  The official name in the documentation is CMI8330, SoundPro
+is the nickname and the big inscription on the chip itself.
+
+The chip emulates a WSS as well as a SB16, but it has certain differences
+in the mixer section which require separate support. It also emulates an
+MPU401 and an OPL3 synthesizer, so you probably want to enable support
+for these, too.
+
+The chip identifies itself as an AD1848, but its mixer is significantly
+more advanced than the original AD1848 one. If your system works with
+either WSS or SB16 and you are having problems with some mixer controls
+(no CD audio, no line-in, etc), you might want to give this driver a try.
+Detection should work, but it hasn't been widely tested, so it might still
+mis-identify the chip. You can still force soundpro=1 in the modprobe
+parameters for ad1848. Please let me know if it happens to you, so I can
+adjust the detection routine.
+
+The chip is capable of doing full-duplex, but since the driver sees it as an
+AD1848, it cannot take advantage of this.  Moreover, the full-duplex mode is
+not achievable through the WSS interface, b/c it needs a dma16 line which is
+assigned only to the SB16 subdevice (with isapnp). Windows documentation
+says the user must use WSS Playback and SB16 Recording for full-duplex, so
+it might be possible to do the same thing under Linux. You can try loading
+up both ad1848 and sb then use one for playback and the other for
+recording. I don't know if this works, b/c I haven't tested it. Anyway, if
+you try it, be very careful: the SB16 mixer *mostly* works, but certain
+settings can have unexpected effects. Use the WSS mixer for best results.
+
+There is also a PCI SoundPro chip. I have not seen this chip, so I have
+no idea if the driver will work with it. I suspect it won't.
+
+As with PnP cards, some configuration is required.  There are two ways
+of doing this.  The most common is to use the isapnptools package to
+initialize the card, and use the kernel module form of the sound
+subsystem and sound drivers.  Alternatively, some BIOS's allow manual
+configuration of installed PnP devices in a BIOS menu, which should
+allow using the non-modular sound drivers, i.e. built into the kernel.
+Since in this latter case you cannot use module parameters, you will
+have to enable support for the SoundPro at compile time.
+
+The IRQ and DMA values can be any that are considered acceptable for a
+WSS.  Assuming you've got isapnp all happy, then you should be able to
+do something like the following (which *must* match the isapnp/BIOS
+configuration):
+
+modprobe ad1848 io=0x530 irq=11 dma=0 soundpro=1
+-and maybe-
+modprobe sb io=0x220 irq=5 dma=1 dma16=5
+
+-then-
+modprobe mpu401 io=0x330 irq=9
+modprobe opl3 io=0x388
+
+If all goes well and you see no error messages, you should be able to
+start using the sound capabilities of your system.  If you get an
+error message while trying to insert the module(s), then make
+sure that the values of the various arguments match what you specified
+in your isapnp configuration file, and that there is no conflict with
+another device for an I/O port or interrupt.  Checking the contents of
+/proc/ioports and /proc/interrupts can be useful to see if you're
+butting heads with another device.
+
+If you do not see the chipset version message, and none of the other
+messages present in the system log are helpful, try adding 'debug=1'
+to the ad1848 parameters, email me the syslog results and I'll do
+my best to help. 
+
+Lastly, if you're using modules and want to set up automatic module
+loading with kmod, the kernel module loader, here is the section I
+currently use in my conf.modules file:
+
+# Sound
+post-install sound modprobe -k ad1848; modprobe -k mpu401; modprobe -k opl3
+options ad1848 io=0x530 irq=11 dma=0
+options sb io=0x220 irq=5 dma=1 dma16=5
+options mpu401 io=0x330 irq=9
+options opl3 io=0x388
+
+The above ensures that ad1848 will be loaded whenever the sound system
+is being used.
+
+Good luck.
+
+Ion
+
+NOT REALLY TESTED:
+- recording
+- recording device selection
+- full-duplex
+
+TODO:
+- implement mixer support for surround, loud, digital CD switches.
+- come up with a scheme which allows recording volumes for each subdevice.
+This is a major OSS API change.
diff --git a/Documentation/sound/oss/Soundblaster b/Documentation/sound/oss/Soundblaster
new file mode 100644
index 00000000..b288d464
--- /dev/null
+++ b/Documentation/sound/oss/Soundblaster
@@ -0,0 +1,53 @@
+modprobe sound
+insmod uart401
+insmod sb ...
+
+This loads the driver for the Sound Blaster and assorted clones. Cards that
+are covered by other drivers should not be using this driver.
+
+The Sound Blaster module takes the following arguments
+
+io		I/O address of the Sound Blaster chip (0x220,0x240,0x260,0x280)
+irq		IRQ of the Sound Blaster chip (5,7,9,10)
+dma		8-bit DMA channel for the Sound Blaster (0,1,3)
+dma16		16-bit DMA channel for SB16 and equivalent cards (5,6,7)
+mpu_io		I/O for MPU chip if present (0x300,0x330)
+
+sm_games=1	Set if you have a Logitech soundman games
+acer=1		Set this to detect cards in some ACER notebooks
+mwave_bug=1	Set if you are trying to use this driver with mwave (see on)
+type		Use this to specify a specific card type
+
+The following arguments are taken if ISAPnP support is compiled in
+
+isapnp=0	Set this to disable ISAPnP detection (use io=0xXXX etc. above)
+multiple=0	Set to disable detection of multiple Soundblaster cards.
+		Consider it a bug if this option is needed, and send in a
+		report.
+pnplegacy=1	Set this to be able to use a PnP card(s) along with a single
+		non-PnP (legacy) card.  Above options for io, irq, etc. are
+		needed, and will apply only to the legacy card.
+reverse=1	Reverses the order of the search in the PnP table.
+uart401=1	Set to enable detection of mpu devices on some clones.
+isapnpjump=n	Jumps to slot n in the driver's PnP table. Use the source,
+		Luke.
+
+You may well want to load the opl3 driver for synth music on most SB and
+clone SB devices
+
+insmod opl3 io=0x388
+
+Using Mwave
+
+To make this driver work with Mwave you must set mwave_bug. You also need
+to warm boot from DOS/Windows with the required firmware loaded under this
+OS. IBM are being difficult about documenting how to load this firmware.
+
+Avance Logic ALS007
+
+This card is supported; see the separate file ALS007 for full details.
+
+Avance Logic ALS100
+
+This card is supported; setup should be as for a standard Sound Blaster 16.
+The driver will identify the audio device as a "Sound Blaster 16 (ALS-100)".
diff --git a/Documentation/sound/oss/Tropez+ b/Documentation/sound/oss/Tropez+
new file mode 100644
index 00000000..b93a6b73
--- /dev/null
+++ b/Documentation/sound/oss/Tropez+
@@ -0,0 +1,26 @@
+From: Paul Barton-Davis 
+
+Here is the configuration I use with a Tropez+ and my modular
+driver:
+
+  alias char-major-14 wavefront
+  alias synth0 wavefront
+  alias mixer0 cs4232
+  alias audio0 cs4232
+  pre-install wavefront modprobe "-k" "cs4232"
+  post-install wavefront modprobe "-k" "opl3"
+  options wavefront io=0x200 irq=9
+  options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
+  options opl3 io=0x388
+
+Things to note: 
+
+       the wavefront options "io" and "irq" ***MUST*** match the "synthio"
+       and "synthirq" cs4232 options.
+
+       you can do without the opl3 module if you don't
+       want to use the OPL/[34] synth on the soundcard
+
+       the opl3 io parameter is conventionally not adjustable.
+
+Please see drivers/sound/README.wavefront for more details.
diff --git a/Documentation/sound/oss/VIBRA16 b/Documentation/sound/oss/VIBRA16
new file mode 100644
index 00000000..68a5a46b
--- /dev/null
+++ b/Documentation/sound/oss/VIBRA16
@@ -0,0 +1,80 @@
+Sound Blaster 16X Vibra addendum
+--------------------------------
+by Marius Ilioaea 
+   Stefan Laudat  
+
+Sat Mar 6 23:55:27 EET 1999
+
+			Hello again,
+	
+	Playing with a SB Vibra 16x soundcard we found it very difficult
+to setup because the kernel reported a lot of DMA errors and wouldn't
+simply play any sound.
+	A good starting point is that the vibra16x chip full-duplex facility
+is neither still exploited by the sb driver found in the linux kernel 
+(tried it with a 2.2.2-ac7), nor in the commercial OSS package (it reports
+it as half-duplex soundcard). Oh, I almost forgot, the RedHat sndconfig
+failed detecting it ;)
+	So, the big problem still remains, because the sb module wants a
+8-bit and a 16-bit dma, which we could not allocate for vibra... it supports
+only two 8-bit dma channels, the second one will be passed to the module
+as a 16 bit channel, the kernel will yield about that but everything will
+be okay, trust us. 
+	The only inconvenient you may find is that you will have
+some sound playing jitters if you have HDD dma support enabled - but this
+will happen with almost all soundcards...
+
+	A fully working isapnp.conf is just here:
+
+
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+# SB 16 and OPL3 devices
+(CONFIGURE CTL00f0/-1 (LD 0
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 3))
+(IO 0 (SIZE 16) (BASE 0x0220))
+(IO 2 (SIZE 4) (BASE 0x0388))
+(NAME "CTL00f0/-1[0]{Audio               }")
+(ACT Y)
+))
+
+# Joystick device - only if you need it :-/
+
+(CONFIGURE CTL00f0/-1 (LD 1
+(IO 0 (SIZE 1) (BASE 0x0200))
+(NAME "CTL00f0/-1[1]{Game                }")
+(ACT Y)
+))
+(WAITFORKEY)
+
+
+
+	So, after a good kernel modules compilation and a 'depmod -a kernel_ver'
+you may want to:
+
+modprobe sb io=0x220 irq=5 dma=1 dma16=3
+
+	Or, take the hard way:
+
+modprobe soundcore
+modprobe sound
+modprobe uart401
+modprobe sb io=0x220 irq=5 dma=1 dma16=3
+# do you need MIDI?
+modprobe opl3=0x388
+
+	Just in case, the kernel sound support should be:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+	
+	Enjoy your new noisy Linux box! ;)
+	
+
diff --git a/Documentation/sound/oss/WaveArtist b/Documentation/sound/oss/WaveArtist
new file mode 100644
index 00000000..f4f3407c
--- /dev/null
+++ b/Documentation/sound/oss/WaveArtist
@@ -0,0 +1,170 @@
+
+ (the following is from the armlinux CVS)
+
+ WaveArtist mixer and volume levels can be accessed via these commands:
+
+  nn30	read registers nn, where nn = 00 - 09 for mixer settings
+		                              0a - 13 for channel volumes
+  mm31	write the volume setting in pairs, where mm = (nn - 10) / 2
+  rr32	write the mixer settings in pairs, where rr = nn/2
+  xx33	reset all settings to default
+  0y34	select mono source, y=0 = left, y=1 = right
+
+                                           bits
+ nn  15  14 13 12 11   10    9     8     7    6     5     4     3     2     1     0
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 00 | 0 | 0  0  1  1 | left line mixer gain       | left aux1 mixer gain        |lmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 01 | 0 | 0  1  0  1 | left aux2 mixer gain       | right 2 left mic gain       |mmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 02 | 0 | 0  1  1  1 | left mic mixer gain        | left mic  | left mixer gain |dith |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 03 | 0 | 1  0  0  1 | left mixer input select                |lrfg | left ADC gain   |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 04 | 0 | 1  0  1  1 | right line mixer gain      | right aux1 mixer gain       |rmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 05 | 0 | 1  1  0  1 | right aux2 mixer gain      | left 2 right mic gain       |test |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 06 | 0 | 1  1  1  1 | right mic mixer gain       | right mic |right mixer gain |rbyps|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 07 | 1 | 0  0  0  1 | right mixer select                     |rrfg | right ADC gain  |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 08 | 1 | 0  0  1  1 | mono mixer gain            |right ADC mux sel|left ADC mux sel |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 09 | 1 | 0  1  0  1 |loopb|left linout|loop|ADCch|TxFch|OffCD|test |loopb|loopb|osamp|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0a | 0 | left PCM channel volume                                                     |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0b | 0 | right PCM channel volume                                                    |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0c | 0 | left FM channel volume                                                      |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0d | 0 | right FM channel volume                                                     |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0e | 0 | left wavetable channel volume                                               |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0f | 0 | right wavetable channel volume                                              |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 10 | 0 | left PCM expansion channel volume                                           |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 11 | 0 | right PCM expansion channel volume                                          |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 12 | 0 | left FM expansion channel volume                                            |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 13 | 0 | right FM expansion channel volume                                           |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+
+ lmute: left mute
+ mmute: mono mute
+ dith: dithds
+ lrfg:
+ rmute: right mute
+ rbyps: right bypass
+ rrfg:
+ ADCch:
+ TxFch:
+ OffCD:
+ osamp:
+
+ And the following diagram is derived from the description in the CVS archive:
+
+ MIC L (mouthpiece)
+   +------+
+ -->PreAmp>-\
+   +--^---+ |
+      |     |
+    r2b4-5  |                                +--------+
+       /----*-------------------------------->5       |
+       |                                     |        |
+       | /----------------------------------->4       |
+       | |                                   |        |
+       | | /--------------------------------->3 1of5  |  +---+
+       | | |                                 |  mux   >-->AMP>--> ADC L
+       | | | /------------------------------->2       |  +-^-+
+       | | | |                               |        |    |
+ Line  | | | | +----+  +------+  +---+  /---->1       |  r3b3-0
+ ------------*->mute>--> Gain >-->   |  |    |        |
+  L    | | |   +----+  +------+  |   |  |  *->0       |
+       | | |                     |   |  |    +---^----+
+ Aux2  | | |   +----+  +------+  |   |  |        |
+ ----------*--->mute>--> Gain >--> M |  |     r8b0-2
+  L    | |     +----+  +------+  |   |  |
+       | |                       |   |  \------\
+ Aux1  | |     +----+  +------+  |   |         |
+ --------*----->mute>--> Gain >--> I |         |
+  L    |       +----+  +------+  |   |         |
+       |                         |   |         |
+       |       +----+  +------+  |   |  +---+  |
+       *------->mute>--> Gain >--> X >-->AMP>--*
+       |       +----+  +------+  |   |  +-^-+  |
+       |                         |   |    |    |
+       |       +----+  +------+  |   |  r2b1-3 |  
+       | /----->mute>--> Gain >--> E |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |         |
+       | | /--->mute>--> Gain >--> R |         |
+       | | |   +----+  +------+  |   |         |
+       | | |                     |   |         |   r9b8-9
+       | | |   +----+  +------+  |   |         |     |
+       | | | /->mute>--> Gain >-->   |         | +---v---+
+       | | | | +----+  +------+  +---+       /-*->0      |
+ DAC   | | | |                               |   |       |
+ ------------*----------------------------------->?      |  +----+
+  L    | | |                                 |   |  Mux  >-->mute>--> L output
+       | | |                                 | /->?      |  +--^-+
+       | | |                                 | | |       |     |
+       | | |                           /--------->?      |   r0b0
+       | | |                           |     | | +-------+
+       | | |                           |     | |
+ Mono  | | |                           |     | | +-------+
+ ----------*                           |     \--->       |  +----+
+       | | |                           |       | |  Mix  >-->mute>--> Mono output
+       | | |                           |       *->       |  +--^-+
+       | | |                           |       | +-------+     |
+       | | |                           |       |             r1b0
+ DAC   | | |                           |       | +-------+
+ ------------*-------------------------*--------->1      |  +----+
+  R    | | | |                                 | |  Mux  >-->mute>--> R output
+       | | | | +----+  +------+  +---+         *->0      |  +--^-+
+       | | | \->mute>--> Gain >-->   |         | +---^---+     |
+       | | |   +----+  +------+  |   |         |     |       r5b0
+       | | |                     |   |         |   r6b0
+       | | |   +----+  +------+  |   |         |
+       | | \--->mute>--> Gain >--> M |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |         |
+       | *----->mute>--> Gain >--> I |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |  +---+  |
+       \------->mute>--> Gain >--> X >-->AMP>--*
+         |     +----+  +------+  |   |  +-^-+  |
+      /--/                       |   |    |    |
+ Aux1 |        +----+  +------+  |   |  r6b1-3 |
+ -------*------>mute>--> Gain >--> E |         |
+  R   | |      +----+  +------+  |   |         |
+      | |                        |   |         |
+ Aux2 | |      +----+  +------+  |   |  /------/
+ ---------*---->mute>--> Gain >--> R |  |
+  R   | | |    +----+  +------+  |   |  |
+      | | |                      |   |  |    +--------+
+ Line | | |    +----+  +------+  |   |  |  *->0       |
+ -----------*-->mute>--> Gain >-->   |  |    |        |
+  R   | | | |  +----+  +------+  +---+  \---->1       |
+      | | | |                                |        |
+      | | | \-------------------------------->2       |  +---+
+      | | |                                  |  Mux   >-->AMP>--> ADC R
+      | | \---------------------------------->3       |  +-^-+
+      | |                                    |        |    |
+      | \------------------------------------>4       |  r7b3-0
+      |                                      |        |
+      \-----*-------------------------------->5       |
+            |                                +---^----+
+    r6b4-5  |                                    |
+      |     |                                  r8b3-5
+   +--v---+ |
+ -->PreAmp>-/
+   +------+
+ MIC R (electret mic)
diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio
new file mode 100644
index 00000000..1a693e69
--- /dev/null
+++ b/Documentation/sound/oss/btaudio
@@ -0,0 +1,92 @@
+
+Intro
+=====
+
+people start bugging me about this with questions, looks like I
+should write up some documentation for this beast.  That way I
+don't have to answer that much mails I hope.  Yes, I'm lazy...
+
+
+You might have noticed that the bt878 grabber cards have actually
+_two_ PCI functions:
+
+$ lspci
+[ ... ]
+00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+[ ... ]
+
+The first does video, it is backward compatible to the bt848.  The second
+does audio.  btaudio is a driver for the second function.  It's a sound
+driver which can be used for recording sound (and _only_ recording, no
+playback).  As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+
+Driver Status
+=============
+
+Still somewhat experimental.  The driver should work stable, i.e. it
+should'nt crash your box.  It might not work as expected, have bugs,
+not being fully OSS API compilant, ...
+
+Latest versions are available from http://bytesex.org/bttv/, the
+driver is in the bttv tarball.  Kernel patches might be available too,
+have a look at http://bytesex.org/bttv/listing.html.
+
+The chip knows two different modes.  btaudio registers two dsp
+devices, one for each mode.  They can not be used at the same time.
+
+
+Digital audio mode
+==================
+
+The chip gives you 16 bit stereo sound.  The sample rate depends on
+the external source which feeds the bt878 with digital sound via I2S
+interface.  There is a insmod option (rate) to tell the driver which
+sample rate the hardware uses (32000 is the default).
+
+One possible source for digital sound is the msp34xx audio processor
+chip which provides digital sound via I2S with 32 kHz sample rate.  My
+Hauppauge board works this way.
+
+The Osprey-200 reportly gives you digital sound with 44100 Hz sample
+rate.  It is also possible that you get no sound at all.
+
+
+analog mode (A/D)
+=================
+
+You can tell the driver to use this mode with the insmod option "analog=1".
+The chip has three analog inputs.  Consequently you'll get a mixer device
+to control these.
+
+The analog mode supports mono only.  Both 8 + 16 bit.  Both are _signed_
+int, which is uncommon for the 8 bit case.  Sample rate range is 119 kHz
+to 448 kHz.  Yes, the number of digits is correct.  The driver supports
+downsampling by powers of two, so you can ask for more usual sample rates
+like 44 kHz too.
+
+With my Hauppauge I get noisy sound on the second input (mapped to line2
+by the mixer device).  Others get a useable signal on line1.
+
+
+some examples
+=============
+
+* read audio data from btaudio (dsp2), send to es1730 (dsp,dsp1):
+  $ sox -w -r 32000 -t ossdsp /dev/dsp2 -t ossdsp /dev/dsp
+
+* read audio data from btaudio, send to esound daemon (which might be
+  running on another host):
+  $ sox -c 2 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -r 32000
+  $ sox -c 1 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -m -r 32000
+
+
+Have fun,
+
+  Gerd
+
+-- 
+Gerd Knorr 
diff --git a/Documentation/sound/oss/mwave b/Documentation/sound/oss/mwave
new file mode 100644
index 00000000..5fbcb160
--- /dev/null
+++ b/Documentation/sound/oss/mwave
@@ -0,0 +1,185 @@
+	How to try to survive an IBM Mwave under Linux SB drivers
+
+
++	IBM have now released documentation of sorts and Torsten is busy
+	trying to make the Mwave work. This is not however a trivial task.
+
+----------------------------------------------------------------------------
+
+OK, first thing - the IRQ problem IS a problem, whether the test is bypassed or
+not.  It is NOT a Linux problem, but an MWAVE problem that is fixed with the
+latest MWAVE patches. So, in other words, don't bypass the test for MWAVES!
+
+I have Windows 95 on /dev/hda1, swap on /dev/hda2, and Red Hat 5 on /dev/hda3.
+
+The steps, then:
+
+    Boot to Linux. 
+    Mount Windows 95 file system (assume mount point = /dos95).
+    mkdir /dos95/linux
+    mkdir /dos95/linux/boot
+    mkdir /dos95/linux/boot/parms
+
+    Copy the kernel, any initrd image, and loadlin to /dos95/linux/boot/.
+
+    Reboot to Windows 95.
+
+    Edit C:/msdos.sys and add or change the following:
+
+        Logo=0
+        BootGUI=0
+
+    Note that msdos.sys is a text file but it needs to be made 'unhidden',
+    readable and writable before it can be edited.  This can be done with
+    DOS' "attrib" command.
+
+    Edit config.sys to have multiple config menus. I have one for windows 95 and
+    five for Linux, like this:
+------------
+[menu]
+menuitem=W95, Windows 95
+menuitem=LINTP, Linux - ThinkPad
+menuitem=LINTP3, Linux - ThinkPad Console
+menuitem=LINDOC, Linux - Docked
+menuitem=LINDOC3, Linux - Docked Console
+menuitem=LIN1, Linux - Single User Mode
+REM menudefault=W95,10
+
+[W95]
+
+[LINTP]
+
+[LINDOC]
+
+[LINTP3]
+
+[LINDOC3]
+
+[LIN1]
+
+[COMMON]
+FILES=30
+REM Please read README.TXT in C:\MWW subdirectory before changing the DOS= statement.
+DOS=HIGH,UMB
+DEVICE=C:\MWW\MANAGER\MWD50430.EXE
+SHELL=c:\command.com /e:2048
+-------------------
+
+The important things are the SHELL and DEVICE statements.
+
+    Then change autoexec.bat. Basically everything in there originally should be
+    done ONLY when Windows 95 is booted. Then you add new things specifically
+    for Linux.  Mine is as follows
+
+---------------
+@ECHO OFF
+if "%CONFIG%" == "W95" goto W95
+
+REM
+REM Linux stuff
+REM
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\MWW\DLL;
+CALL MWAVE START NOSHOW
+c:\linux\boot\loadlin.exe @c:\linux\boot\parms\%CONFIG%.par
+
+:W95
+REM
+REM Windows 95 stuff
+REM
+c:\toolkit\guard
+SET MSINPUT=C:\MSINPUT
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+REM The following is used by DOS games to recognize Sound Blaster hardware.
+REM If hardware settings are changed, please change this line as well.
+REM See the Mwave README file for instructions.
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\WINDOWS\COMMAND;E:\ORAWIN95\BIN;f:\msdev\bin;e:\v30\bin.dbg;v:\devt\v30\bin;c:\JavaSDK\Bin;C:\MWW\DLL;
+SET INCLUDE=f:\MSDEV\INCLUDE;F:\MSDEV\MFC\INCLUDE
+SET LIB=F:\MSDEV\LIB;F:\MSDEV\MFC\LIB
+win
+
+------------------------
+
+Now build a file in c:\linux\boot\parms for each Linux config that you have.
+
+For example, my LINDOC3 config is for a docked Thinkpad at runlevel 3 with no
+initrd image, and has a parameter file named LINDOC3.PAR in c:\linux\boot\parms:
+
+-----------------------
+#   LOADLIN @param_file image=other_image root=/dev/other
+#
+#	Linux Console in docking station
+#
+c:\linux\boot\zImage.krn        # First value must be filename of Linux kernel.
+root=/dev/hda3                  # device which gets mounted as root FS
+ro                              # Other kernel arguments go here.
+apm=off
+doc=yes
+3
+-----------------------
+
+The doc=yes parameter is an environment variable used by my init scripts, not
+a kernel argument.
+
+However, the apm=off parameter IS a kernel argument!  APM, at least in my setup,
+causes the kernel to crash when loaded via loadlin (but NOT when loaded via
+LILO). The APM stuff COULD be forced out of the kernel via the kernel compile
+options. Instead, I got an unofficial patch to the APM drivers that allows them
+to be dynamically deactivated via kernel arguments. Whatever you chose to
+document, APM, it seems, MUST be off for setups like mine.
+
+Now make sure C:\MWW\MWCONFIG.REF looks like this:
+
+----------------------
+[NativeDOS]
+Default=SB1.5
+SBInputSource=CD
+SYNTH=FM
+QSound=OFF
+Reverb=OFF
+Chorus=OFF
+ReverbDepth=5
+ChorusDepth=5
+SBInputVolume=5
+SBMainVolume=10
+SBWaveVolume=10
+SBSynthVolume=10
+WaveTableVolume=10
+AudioPowerDriver=ON
+
+[FastCFG]
+Show=No
+HideOption=Off
+-----------------------------
+
+OR the Default= line COULD be
+
+Default=SBPRO
+
+Reboot to Windows 95 and choose Linux. When booted, use sndconfig to configure
+the sound modules and voilà - ThinkPad sound with Linux.
+
+Now the gotchas - you can either have CD sound OR Mixers but not both. That's a
+problem with the SB1.5 (CD sound) or SBPRO (Mixers) settings. No one knows why
+this is!
+
+For some reason MPEG3 files, when played through mpg123, sound like they
+are playing at 1/8th speed - not very useful!  If you have ANY insight
+on why this second thing might be happening, I would be grateful.
+
+===========================================================
+    _/      _/_/_/_/
+   _/_/  _/_/     _/
+  _/ _/_/ _/_/_/_/      Martin John Bartlett
+ _/  _/  _/     _/      (martin@nitram.demon.co.uk)
+_/      _/_/_/_/
+       _/
+_/    _/
+ _/_/
+===========================================================
diff --git a/Documentation/sound/oss/oss-parameters.txt b/Documentation/sound/oss/oss-parameters.txt
new file mode 100644
index 00000000..3ab391e7
--- /dev/null
+++ b/Documentation/sound/oss/oss-parameters.txt
@@ -0,0 +1,51 @@
+                          OSS Kernel Parameters
+                          ~~~~~~~~~~~~~~~~~~~~~
+
+See Documentation/kernel-parameters.txt for general information on
+specifying module parameters.
+
+This document may not be entirely up to date and comprehensive. The command
+"modinfo -p ${modulename}" shows a current list of all parameters of a loadable
+module. Loadable modules, after being loaded into the running kernel, also
+reveal their parameters in /sys/module/${modulename}/parameters/. Some of these
+parameters may be changed at runtime by the command
+"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}".
+
+
+	ad1848=		[HW,OSS]
+			Format: ,,,,
+
+	aedsp16=	[HW,OSS] Audio Excel DSP 16
+			Format: ,,,,,
+			See also header of sound/oss/aedsp16.c.
+
+	dmasound=	[HW,OSS] Sound subsystem buffers
+
+	mpu401=		[HW,OSS]
+			Format: ,
+
+	opl3=		[HW,OSS]
+			Format: 
+
+	pas2=		[HW,OSS] Format:
+			,,,,,,,
+
+	pss=		[HW,OSS] Personal Sound System (ECHO ESC614)
+			Format:
+			,,,,,
+
+	sscape=		[HW,OSS]
+			Format: ,,,,
+
+	trix=		[HW,OSS] MediaTrix AudioTrix Pro
+			Format:
+			,,,,,,,,
+
+	uart401=	[HW,OSS]
+			Format: ,
+
+	uart6850=	[HW,OSS]
+			Format: ,
+
+	waveartist=	[HW,OSS]
+			Format: ,,,
diff --git a/Documentation/sound/oss/ultrasound b/Documentation/sound/oss/ultrasound
new file mode 100644
index 00000000..eed331c7
--- /dev/null
+++ b/Documentation/sound/oss/ultrasound
@@ -0,0 +1,30 @@
+modprobe sound
+insmod ad1848
+insmod gus io=* irq=* dma=* ...
+
+This loads the driver for the Gravis Ultrasound family of sound cards.
+
+The gus module takes the following arguments
+
+io		I/O address of the Ultrasound card (eg. io=0x220)
+irq		IRQ of the Sound Blaster card 
+dma     	DMA channel for the Sound Blaster
+dma16   	2nd DMA channel, only needed for full duplex operation
+type		1 for PnP card
+gus16		1 for using 16 bit sampling daughter board
+no_wave_dma	Set to disable DMA usage for wavetable (see note)
+db16		???
+
+
+no_wave_dma option
+
+This option defaults to a value of 0, which allows the Ultrasound wavetable
+DSP to use DMA for playback and downloading samples. This is the same
+as the old behaviour. If set to 1, no DMA is needed for downloading samples,
+and allows owners of a GUS MAX to make use of simultaneous digital audio
+(/dev/dsp), MIDI, and wavetable playback.
+
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
diff --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd
new file mode 100644
index 00000000..4c6cbdb3
--- /dev/null
+++ b/Documentation/sound/oss/vwsnd
@@ -0,0 +1,293 @@
+vwsnd - Sound driver for the Silicon Graphics 320 and 540 Visual
+Workstations' onboard audio.
+
+Copyright 1999 Silicon Graphics, Inc.  All rights reserved.
+
+
+At the time of this writing, March 1999, there are two models of
+Visual Workstation, the 320 and the 540.  This document only describes
+those models.  Future Visual Workstation models may have different
+sound capabilities, and this driver will probably not work on those
+boxes.
+
+The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio
+codec chip.  The AD1843 is accessed through the Cobalt I/O ASIC, also
+known as Lithium.  This driver programs both chips.
+
+==============================================================================
+QUICK CONFIGURATION
+
+	# insmod soundcore
+	# insmod vwsnd
+
+==============================================================================
+I/O CONNECTIONS
+
+On the Visual Workstation, only three of the AD1843 inputs are hooked
+up.  The analog line in jacks are connected to the AD1843's AUX1
+input.  The CD audio lines are connected to the AD1843's AUX2 input.
+The microphone jack is connected to the AD1843's MIC input.  The mic
+jack is mono, but the signal is delivered to both the left and right
+MIC inputs.  You can record in stereo from the mic input, but you will
+get the same signal on both channels (within the limits of A/D
+accuracy).  Full scale on the Line input is +/- 2.0 V.  Full scale on
+the MIC input is 20 dB less, or +/- 0.2 V.
+
+The AD1843's LOUT1 outputs are connected to the Line Out jacks.  The
+AD1843's HPOUT outputs are connected to the speaker/headphone jack.
+LOUT2 is not connected.  Line out's maximum level is +/- 2.0 V peak to
+peak.  The speaker/headphone out's maximum is +/- 4.0 V peak to peak.
+
+The AD1843's PCM input channel and one of its output channels (DAC1)
+are connected to Lithium.  The other output channel (DAC2) is not
+connected.
+
+==============================================================================
+CAPABILITIES
+
+The AD1843 has PCM input and output (Pulse Code Modulation, also known
+as wavetable).  PCM input and output can be mono or stereo in any of
+four formats.  The formats are 16 bit signed and 8 bit unsigned,
+u-Law, and A-Law format.  Any sample rate from 4 KHz to 49 KHz is
+available, in 1 Hz increments.
+
+The AD1843 includes an analog mixer that can mix all three input
+signals (line, mic and CD) into the analog outputs.  The mixer has a
+separate gain control and mute switch for each input.
+
+There are two outputs, line out and speaker/headphone out.  They
+always produce the same signal, and the speaker always has 3 dB more
+gain than the line out.  The speaker/headphone output can be muted,
+but this driver does not export that function.
+
+The hardware can sync audio to the video clock, but this driver does
+not have a way to specify syncing to video.
+
+==============================================================================
+PROGRAMMING
+
+This section explains the API supported by the driver.  Also see the
+Open Sound Programming Guide at http://www.opensound.com/pguide/ .
+This section assumes familiarity with that document.
+
+The driver has two interfaces, an I/O interface and a mixer interface.
+There is no MIDI or sequencer capability.
+
+==============================================================================
+PROGRAMMING PCM I/O
+
+The I/O interface is usually accessed as /dev/audio or /dev/dsp.
+Using the standard Open Sound System (OSS) ioctl calls, the sample
+rate, number of channels, and sample format may be set within the
+limitations described above.  The driver supports triggering.  It also
+supports getting the input and output pointers with one-sample
+accuracy.
+
+The SNDCTL_DSP_GETCAP ioctl returns these capabilities.
+
+	DSP_CAP_DUPLEX - driver supports full duplex.
+
+	DSP_CAP_TRIGGER - driver supports triggering.
+
+	DSP_CAP_REALTIME - values returned by SNDCTL_DSP_GETIPTR
+	and SNDCTL_DSP_GETOPTR are accurate to a few samples.
+
+Memory mapping (mmap) is not implemented.
+
+The driver permits subdivided fragment sizes from 64 to 4096 bytes.
+The number of fragments can be anything from 3 fragments to however
+many fragments fit into 124 kilobytes.  It is up to the user to
+determine how few/small fragments can be used without introducing
+glitches with a given workload.  Linux is not realtime, so we can't
+promise anything.  (sigh...)
+
+When this driver is switched into or out of mu-Law or A-Law mode on
+output, it may produce an audible click.  This is unavoidable.  To
+prevent clicking, use signed 16-bit mode instead, and convert from
+mu-Law or A-Law format in software.
+
+==============================================================================
+PROGRAMMING THE MIXER INTERFACE
+
+The mixer interface is usually accessed as /dev/mixer.  It is accessed
+through ioctls.  The mixer allows the application to control gain or
+mute several audio signal paths, and also allows selection of the
+recording source.
+
+Each of the constants described here can be read using the
+MIXER_READ(SOUND_MIXER_xxx) ioctl.  Those that are not read-only can
+also be written using the MIXER_WRITE(SOUND_MIXER_xxx) ioctl.  In most
+cases,  defines constants SOUND_MIXER_READ_xxx and
+SOUND_MIXER_WRITE_xxx which work just as well.
+
+SOUND_MIXER_CAPS	Read-only
+
+This is a mask of optional driver capabilities that are implemented.
+This driver's only capability is SOUND_CAP_EXCL_INPUT, which means
+that only one recording source can be active at a time.
+
+SOUND_MIXER_DEVMASK	Read-only
+
+This is a mask of the sound channels.  This driver's channels are PCM,
+LINE, MIC, CD, and RECLEV.
+
+SOUND_MIXER_STEREODEVS	Read-only
+
+This is a mask of which sound channels are capable of stereo.  All
+channels are capable of stereo.  (But see caveat on MIC input in I/O
+CONNECTIONS section above).
+
+SOUND_MIXER_OUTMASK	Read-only
+
+This is a mask of channels that route inputs through to outputs.
+Those are LINE, MIC, and CD.
+
+SOUND_MIXER_RECMASK	Read-only
+
+This is a mask of channels that can be recording sources.  Those are
+PCM, LINE, MIC, CD.
+
+SOUND_MIXER_PCM		Default: 0x5757 (0 dB)
+
+This is the gain control for PCM output.  The left and right channel
+gain are controlled independently.  This gain control has 64 levels,
+which range from -82.5 dB to +12.0 dB in 1.5 dB steps.  Those 64
+levels are mapped onto 100 levels at the ioctl, see below.
+
+SOUND_MIXER_LINE	Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the Line In source into the
+outputs.  The left and right channel gain are controlled
+independently.  This gain control has 32 levels, which range from
+-34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
+100 levels at the ioctl, see below.
+
+SOUND_MIXER_MIC		Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the MIC source into the outputs.
+The left and right channel gain are controlled independently.  This
+gain control has 32 levels, which range from -34.5 dB to +12.0 dB in
+1.5 dB steps.  Those 32 levels are mapped onto 100 levels at the
+ioctl, see below.
+
+SOUND_MIXER_CD		Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the CD audio source into the
+outputs.  The left and right channel gain are controlled
+independently.  This gain control has 32 levels, which range from
+-34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
+100 levels at the ioctl, see below.
+
+SOUND_MIXER_RECLEV	 Default: 0 (0 dB)
+
+This is the gain control for PCM input (RECording LEVel).  The left
+and right channel gain are controlled independently.  This gain
+control has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dB
+steps.  Those 16 levels are mapped onto 100 levels at the ioctl, see
+below.
+
+SOUND_MIXER_RECSRC	 Default: SOUND_MASK_LINE
+
+This is a mask of currently selected PCM input sources (RECording
+SouRCes).  Because the AD1843 can only have a single recording source
+at a time, only one bit at a time can be set in this mask.  The
+allowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC,
+or SOUND_MASK_CD.  Selecting SOUND_MASK_PCM sets up internal
+resampling which is useful for loopback testing and for hardware
+sample rate conversion.  But software sample rate conversion is
+probably faster, so I don't know how useful that is.
+
+SOUND_MIXER_OUTSRC	DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CD
+
+This is a mask of sources that are currently passed through to the
+outputs.  Those sources whose bits are not set are muted.
+
+==============================================================================
+GAIN CONTROL
+
+There are five gain controls listed above.  Each has 16, 32, or 64
+steps.  Each control has 1.5 dB of gain per step.  Each control is
+stereo.
+
+The OSS defines the argument to a channel gain ioctl as having two
+components, left and right, each of which ranges from 0 to 100.  The
+two components are packed into the same word, with the left side gain
+in the least significant byte, and the right side gain in the second
+least significant byte.  In C, we would say this.
+
+	#include 
+
+	...
+
+	 	assert(leftgain >= 0 && leftgain <= 100);
+		assert(rightgain >= 0 && rightgain <= 100);
+		arg = leftgain | rightgain << 8;
+
+So each OSS gain control has 101 steps.  But the hardware has 16, 32,
+or 64 steps.  The hardware steps are spread across the 101 OSS steps
+nearly evenly.  The conversion formulas are like this, given N equals
+16, 32, or 64.
+
+	int round = N/2 - 1;
+	OSS_gain_steps = (hw_gain_steps * 100 + round) / (N - 1);
+	hw_gain_steps = (OSS_gain_steps * (N - 1) + round) / 100;
+
+Here is a snippet of C code that will return the left and right gain
+of any channel in dB.  Pass it one of the predefined gain_desc_t
+structures to access any of the five channels' gains.
+
+	typedef struct gain_desc {
+		float min_gain;
+		float gain_step;
+		int nbits;
+		int chan;
+	} gain_desc_t;
+
+	const gain_desc_t gain_pcm    = { -82.5, 1.5, 6, SOUND_MIXER_PCM    };
+	const gain_desc_t gain_line   = { -34.5, 1.5, 5, SOUND_MIXER_LINE   };
+	const gain_desc_t gain_mic    = { -34.5, 1.5, 5, SOUND_MIXER_MIC    };
+	const gain_desc_t gain_cd     = { -34.5, 1.5, 5, SOUND_MIXER_CD     };
+	const gain_desc_t gain_reclev = {   0.0, 1.5, 4, SOUND_MIXER_RECLEV };
+
+	int get_gain_dB(int fd, const gain_desc_t *gp,
+			float *left, float *right)
+	{
+		int word;
+		int lg, rg;
+		int mask = (1 << gp->nbits) - 1;
+
+		if (ioctl(fd, MIXER_READ(gp->chan), &word) != 0)
+			return -1;	/* fail */
+		lg = word & 0xFF;
+		rg = word >> 8 & 0xFF;
+		lg = (lg * mask + mask / 2) / 100;
+		rg = (rg * mask + mask / 2) / 100;
+		*left = gp->min_gain + gp->gain_step * lg;
+		*right = gp->min_gain + gp->gain_step * rg;
+		return 0;
+	}	
+
+And here is the corresponding routine to set a channel's gain in dB.
+
+	int set_gain_dB(int fd, const gain_desc_t *gp, float left, float right)
+	{
+		float max_gain =
+			gp->min_gain + (1 << gp->nbits) * gp->gain_step;
+		float round = gp->gain_step / 2;
+		int mask = (1 << gp->nbits) - 1;
+		int word;
+		int lg, rg;
+
+		if (left < gp->min_gain || right < gp->min_gain)
+			return EINVAL;
+		lg = (left - gp->min_gain + round) / gp->gain_step;
+		rg = (right - gp->min_gain + round) / gp->gain_step;
+		if (lg >= (1 << gp->nbits) || rg >= (1 << gp->nbits))
+			return EINVAL;
+		lg = (100 * lg + mask / 2) / mask;
+		rg = (100 * rg + mask / 2) / mask;
+		word = lg | rg << 8;
+
+		return ioctl(fd, MIXER_WRITE(gp->chan), &word);
+	}
+
-- 
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