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Diffstat (limited to 'Documentation/sound')
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diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt deleted file mode 100644 index 221b81016db..00000000000 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ /dev/null @@ -1,2319 +0,0 @@ - - Advanced Linux Sound Architecture - Driver - ========================================== - Configuration guide - - -Kernel Configuration -==================== - -To enable ALSA support you need at least to build the kernel with -primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS, -you don't have to choose any of the OSS modules. - -Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and -PCM supports if you want to run OSS applications with ALSA. - -If you want to support the WaveTable functionality on cards such as -SB Live! then you need to enable "Sequencer support" -(CONFIG_SND_SEQUENCER). - -To make ALSA debug messages more verbose, enable the "Verbose printk" -and "Debug" options. To check for memory leaks, turn on "Debug memory" -too. "Debug detection" will add checks for the detection of cards. - -Please note that all the ALSA ISA drivers support the Linux isapnp API -(if the card supports ISA PnP). You don't need to configure the cards -using isapnptools. - - -Creating ALSA devices -===================== - -This depends on your distribution, but normally you use the /dev/MAKEDEV -script to create the necessary device nodes. On some systems you use a -script named 'snddevices'. - - -Module parameters -================= - -The user can load modules with options. If the module supports more than -one card and you have more than one card of the same type then you can -specify multiple values for the option separated by commas. - -Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. - - Module snd - ---------- - - The core ALSA module. It is used by all ALSA card drivers. - It takes the following options which have global effects. - - major - major number for sound driver - - Default: 116 - cards_limit - - limiting card index for auto-loading (1-8) - - Default: 1 - - For auto-loading more than one card, specify this - option together with snd-card-X aliases. - slots - Reserve the slot index for the given driver. - This option takes multiple strings. - See "Module Autoloading Support" section for details. - debug - Specifies the debug message level - (0 = disable debug prints, 1 = normal debug messages, - 2 = verbose debug messages) - This option appears only when CONFIG_SND_DEBUG=y. - This option can be dynamically changed via sysfs - /sys/modules/snd/parameters/debug file. - - Module snd-pcm-oss - ------------------ - - The PCM OSS emulation module. - This module takes options which change the mapping of devices. - - dsp_map - PCM device number maps assigned to the 1st OSS device. - - Default: 0 - adsp_map - PCM device number maps assigned to the 2st OSS device. - - Default: 1 - nonblock_open - - Don't block opening busy PCM devices. Default: 1 - - For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of - the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped - to PCM #0 of the card #0. - For changing the second or later card, specify the option with - commas, such like "dsp_map=0,1". - - nonblock_open option is used to change the behavior of the PCM - regarding opening the device. When this option is non-zero, - opening a busy OSS PCM device won't be blocked but return - immediately with EAGAIN (just like O_NONBLOCK flag). - - Module snd-rawmidi - ------------------ - - This module takes options which change the mapping of devices. - similar to those of the snd-pcm-oss module. - - midi_map - MIDI device number maps assigned to the 1st OSS device. - - Default: 0 - amidi_map - MIDI device number maps assigned to the 2st OSS device. - - Default: 1 - - Common parameters for top sound card modules - -------------------------------------------- - - Each of top level sound card module takes the following options. - - index - index (slot #) of sound card - - Values: 0 through 31 or negative - - If nonnegative, assign that index number - - if negative, interpret as a bitmask of permissible - indices; the first free permitted index is assigned - - Default: -1 - id - card ID (identifier or name) - - Can be up to 15 characters long - - Default: the card type - - A directory by this name is created under /proc/asound/ - containing information about the card - - This ID can be used instead of the index number in - identifying the card - enable - enable card - - Default: enabled, for PCI and ISA PnP cards - - Module snd-adlib - ---------------- - - Module for AdLib FM cards. - - port - port # for OPL chip - - This module supports multiple cards. It does not support autoprobe, so - the port must be specified. For actual AdLib FM cards it will be 0x388. - Note that this card does not have PCM support and no mixer; only FM - synthesis. - - Make sure you have "sbiload" from the alsa-tools package available and, - after loading the module, find out the assigned ALSA sequencer port - number through "sbiload -l". Example output: - - Port Client name Port name - 64:0 OPL2 FM synth OPL2 FM Port - - Load the std.sb and drums.sb patches also supplied by sbiload: - - sbiload -p 64:0 std.sb drums.sb - - If you use this driver to drive an OPL3, you can use std.o3 and drums.o3 - instead. To have the card produce sound, use aplaymidi from alsa-utils: - - aplaymidi -p 64:0 foo.mid - - Module snd-ad1816a - ------------------ - - Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. - - clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) - - This module supports multiple cards, autoprobe and PnP. - - Module snd-ad1848 - ----------------- - - Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. - - port - port # for AD1848 chip - irq - IRQ # for AD1848 chip - dma1 - DMA # for AD1848 chip (0,1,3) - - This module supports multiple cards. It does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-ad1889 - ----------------- - - Module for Analog Devices AD1889 chips. - - ac97_quirk - AC'97 workaround for strange hardware - See the description of intel8x0 module for details. - - This module supports multiple cards. - - Module snd-ali5451 - ------------------ - - Module for ALi M5451 PCI chip. - - pcm_channels - Number of hardware channels assigned for PCM - spdif - Support SPDIF I/O - - Default: disabled - - This module supports one chip and autoprobe. - - The power-management is supported. - - Module snd-als100 - ----------------- - - Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-als300 - ----------------- - - Module for Avance Logic ALS300 and ALS300+ - - This module supports multiple cards. - - The power-management is supported. - - Module snd-als4000 - ------------------ - - Module for sound cards based on Avance Logic ALS4000 PCI chip. - - joystick_port - port # for legacy joystick support. - 0 = disabled (default), 1 = auto-detect - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-asihpi - ----------------- - - Module for AudioScience ASI soundcards - - enable_hpi_hwdep - enable HPI hwdep for AudioScience soundcard - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-atiixp - ----------------- - - Module for ATI IXP 150/200/250/400 AC97 controllers. - - ac97_clock - AC'97 clock (default = 48000) - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - ac97_codec - Workaround to specify which AC'97 codec - instead of probing. If this works for you - file a bug with your `lspci -vn` output. - -2 -- Force probing. - -1 -- Default behavior. - 0-2 -- Use the specified codec. - spdif_aclink - S/PDIF transfer over AC-link (default = 1) - - This module supports one card and autoprobe. - - ATI IXP has two different methods to control SPDIF output. One is - over AC-link and another is over the "direct" SPDIF output. The - implementation depends on the motherboard, and you'll need to - choose the correct one via spdif_aclink module option. - - The power-management is supported. - - Module snd-atiixp-modem - ----------------------- - - Module for ATI IXP 150/200/250 AC97 modem controllers. - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-au8810, snd-au8820, snd-au8830 - ----------------------------------------- - - Module for Aureal Vortex, Vortex2 and Advantage device. - - pcifix - Control PCI workarounds - 0 = Disable all workarounds - 1 = Force the PCI latency of the Aureal card to 0xff - 2 = Force the Extend PCI#2 Internal Master for Efficient - Handling of Dummy Requests on the VIA KT133 AGP Bridge - 3 = Force both settings - 255 = Autodetect what is required (default) - - This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware - EQ, mpu401, gameport. A3D and wavetable support are still in development. - Development and reverse engineering work is being coordinated at - http://savannah.nongnu.org/projects/openvortex/ - SPDIF output has a copy of the AC97 codec output, unless you use the - "spdif" pcm device, which allows raw data passthru. - The hardware EQ hardware and SPDIF is only present in the Vortex2 and - Advantage. - - Note: Some ALSA mixer applications don't handle the SPDIF sample rate - control correctly. If you have problems regarding this, try - another ALSA compliant mixer (alsamixer works). - - Module snd-azt1605 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-azt2316 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-aw2 - -------------- - - Module for Audiowerk2 sound card - - This module supports multiple cards. - - Module snd-azt2320 - ------------------ - - Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). - - This module supports multiple cards, PnP and autoprobe. - - The power-management is supported. - - Module snd-azt3328 - ------------------ - - Module for sound cards based on Aztech AZF3328 PCI chip. - - joystick - Enable joystick (default off) - - This module supports multiple cards. - - Module snd-bt87x - ---------------- - - Module for video cards based on Bt87x chips. - - digital_rate - Override the default digital rate (Hz) - load_all - Load the driver even if the card model isn't known - - This module supports multiple cards. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - Module snd-ca0106 - ----------------- - - Module for Creative Audigy LS and SB Live 24bit - - This module supports multiple cards. - - - Module snd-cmi8330 - ------------------ - - Module for sound cards based on C-Media CMI8330 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - wssport - port # for CMI8330 chip (WSS) - wssirq - IRQ # for CMI8330 chip (WSS) - wssdma - first DMA # for CMI8330 chip (WSS) - sbport - port # for CMI8330 chip (SB16) - sbirq - IRQ # for CMI8330 chip (SB16) - sbdma8 - 8bit DMA # for CMI8330 chip (SB16) - sbdma16 - 16bit DMA # for CMI8330 chip (SB16) - fmport - (optional) OPL3 I/O port - mpuport - (optional) MPU401 I/O port - mpuirq - (optional) MPU401 irq # - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-cmipci - ----------------- - - Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. - - mpu_port - port address of MIDI interface (8338 only): - 0x300,0x310,0x320,0x330 = legacy port, - 0 = disable (default) - fm_port - port address of OPL-3 FM synthesizer (8x38 only): - 0x388 = legacy port, - 1 = integrated PCI port (default on 8738), - 0 = disable - soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) - (default = 1) - joystick_port - Joystick port address (0 = disable, 1 = auto-detect) - - This module supports autoprobe and multiple cards. - - The power-management is supported. - - Module snd-cs4231 - ----------------- - - Module for sound cards based on CS4231 ISA chips. - - port - port # for CS4231 chip - mpu_port - port # for MPU-401 UART (optional), -1 = disable - irq - IRQ # for CS4231 chip - mpu_irq - IRQ # for MPU-401 UART - dma1 - first DMA # for CS4231 chip - dma2 - second DMA # for CS4231 chip - - This module supports multiple cards. This module does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-cs4236 - ----------------- - - Module for sound cards based on CS4232/CS4232A, - CS4235/CS4236/CS4236B/CS4237B/ - CS4238B/CS4239 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for CS4236 chip (PnP setup - 0x534) - cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) - mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable - fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable - irq - IRQ # for CS4236 chip (5,7,9,11,12,15) - mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) - dma1 - first DMA # for CS4236 chip (0,1,3) - dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable - - This module supports multiple cards. This module does not support autoprobe - (if ISA PnP is not used) thus main port and control port must be - specified!!! Other ports are optional. - - The power-management is supported. - - This module is aliased as snd-cs4232 since it provides the old - snd-cs4232 functionality, too. - - Module snd-cs4281 - ----------------- - - Module for Cirrus Logic CS4281 soundchip. - - dual_codec - Secondary codec ID (0 = disable, default) - - This module supports multiple cards. - - The power-management is supported. - - Module snd-cs46xx - ----------------- - - Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ - CS4624/CS4630/CS4280 PCI chips. - - external_amp - Force to enable external amplifier. - thinkpad - Force to enable Thinkpad's CLKRUN control. - mmap_valid - Support OSS mmap mode (default = 0). - - This module supports multiple cards and autoprobe. - Usually external amp and CLKRUN controls are detected automatically - from PCI sub vendor/device ids. If they don't work, give the options - above explicitly. - - The power-management is supported. - - Module snd-cs5530 - _________________ - - Module for Cyrix/NatSemi Geode 5530 chip. - - Module snd-cs5535audio - ---------------------- - - Module for multifunction CS5535 companion PCI device - - The power-management is supported. - - Module snd-ctxfi - ---------------- - - Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) - * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series - * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series - * Creative Sound Blaster X-Fi Titanium Professional Audio - * Creative Sound Blaster X-Fi Titanium - * Creative Sound Blaster X-Fi Elite Pro - * Creative Sound Blaster X-Fi Platinum - * Creative Sound Blaster X-Fi Fatal1ty - * Creative Sound Blaster X-Fi XtremeGamer - * Creative Sound Blaster X-Fi XtremeMusic - - reference_rate - reference sample rate, 44100 or 48000 (default) - multiple - multiple to ref. sample rate, 1 or 2 (default) - subsystem - override the PCI SSID for probing; the value - consists of SSVID << 16 | SSDID. The default is - zero, which means no override. - - This module supports multiple cards. - - Module snd-darla20 - ------------------ - - Module for Echoaudio Darla20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-darla24 - ------------------ - - Module for Echoaudio Darla24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-dt019x - ----------------- - - Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP - only) - - This module supports multiple cards. This module is enabled only with - ISA PnP support. - - The power-management is supported. - - Module snd-dummy - ---------------- - - Module for the dummy sound card. This "card" doesn't do any output - or input, but you may use this module for any application which - requires a sound card (like RealPlayer). - - pcm_devs - Number of PCM devices assigned to each card - (default = 1, up to 4) - pcm_substreams - Number of PCM substreams assigned to each PCM - (default = 8, up to 128) - hrtimer - Use hrtimer (=1, default) or system timer (=0) - fake_buffer - Fake buffer allocations (default = 1) - - When multiple PCM devices are created, snd-dummy gives different - behavior to each PCM device: - 0 = interleaved with mmap support - 1 = non-interleaved with mmap support - 2 = interleaved without mmap - 3 = non-interleaved without mmap - - As default, snd-dummy drivers doesn't allocate the real buffers - but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the 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, - 3 = VIACOMBO, 4 = COMBO) - 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) - align_buffer_size - Force rounding of buffer/period sizes to multiples - of 128 bytes. This is more efficient in terms of memory - access but isn't required by the HDA spec and prevents - users from specifying exact period/buffer sizes. - (default = on) - snoop - Enable/disable snooping (default = on) - - This module supports multiple cards and autoprobe. - - See Documentation/sound/alsa/HD-Audio.txt for more details about - HD-audio driver. - - Each codec may have a model table for different configurations. - If your machine isn't listed there, the default (usually minimal) - configuration is set up. You can pass "model=<name>" option to - specify a certain model in such a case. There are different - models depending on the codec chip. The list of available models - is found in HD-Audio-Models.txt - - The model name "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.) - - position_fix=3 is specific to VIA devices. The position - of the capture stream is checked from both LPIB and POSBUF - values. position_fix=4 is a combination mode, using LPIB - for playback and POSBUF for capture. - - NB: If you get many "azx_get_response timeout" messages at - loading, it's likely a problem of interrupts (e.g. ACPI irq - routing). Try to boot with options like "pci=noacpi". Also, you - can try "single_cmd=1" module option. This will switch the - communication method between HDA controller and codecs to the - single immediate commands instead of CORB/RIRB. Basically, the - single command mode is provided only for BIOS, and you won't get - unsolicited events, too. But, at least, this works independently - from the irq. Remember this is a last resort, and should be - avoided as much as possible... - - MORE NOTES ON "azx_get_response timeout" PROBLEMS: - On some 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/DGX - * AuzenTech X-Meridian - * AuzenTech X-Meridian 2G - * Bgears b-Enspirer - * Club3D Theatron DTS - * HT-Omega Claro (plus) - * HT-Omega Claro halo (XT) - * Kuroutoshikou CMI8787-HG2PCI - * Razer Barracuda AC-1 - * Sondigo Inferno - * TempoTec HiFier Fantasia - * TempoTec HiFier Serenade - - This module supports autoprobe and multiple cards. - - Module snd-pcsp - ----------------- - - Module for internal PC-Speaker. - - nopcm - Disable PC-Speaker PCM sound. Only beeps remain. - nforce_wa - enable NForce chipset workaround. Expect bad sound. - - This module supports system beeps, some kind of PCM playback and - even a few mixer controls. - - Module snd-pcxhr - ---------------- - - Module for Digigram PCXHR boards - - This module supports multiple cards. - - Module snd-portman2x4 - --------------------- - - Module for Midiman Portman 2x4 parallel port MIDI interface - - This module supports multiple cards. - - Module snd-powermac (on ppc only) - --------------------------------- - - Module for PowerMac, iMac and iBook on-board soundchips - - enable_beep - enable beep using PCM (enabled as default) - - Module supports autoprobe a chip. - - Note: the driver may have problems regarding endianness. - - The power-management is supported. - - Module snd-pxa2xx-ac97 (on arm only) - ------------------------------------ - - Module for AC97 driver for the Intel PXA2xx chip - - For ARM architecture only. - - The power-management is supported. - - Module snd-riptide - ------------------ - - Module for Conexant Riptide chip - - joystick_port - Joystick port # (default: 0x200) - mpu_port - MPU401 port # (default: 0x330) - opl3_port - OPL3 port # (default: 0x388) - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - You need to install the firmware file "riptide.hex" to the standard - firmware path (e.g. /lib/firmware). - - Module snd-rme32 - ---------------- - - Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, - Prodif96 and Prodif Gold) sound cards. - - This module supports multiple cards. - - Module snd-rme96 - ---------------- - - Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. - - This module supports multiple cards. - - Module snd-rme9652 - ------------------ - - Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. - - precise_ptr - Enable precise pointer (doesn't work reliably). - (default = 0) - - This module supports multiple cards. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any RME9652 cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-sa11xx-uda1341 (on arm only) - --------------------------------------- - - Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. - - Module supports only one card. - Module has no enable and index options. - - The power-management is supported. - - Module snd-sb8 - -------------- - - Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, - SoundBlaster 2.0, - SoundBlaster Pro - - port - port # for SB DSP chip (0x220,0x240,0x260) - irq - IRQ # for SB DSP chip (5,7,9,10) - dma8 - DMA # for SB DSP chip (1,3) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-sb16 and snd-sbawe - ----------------------------- - - Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), - SoundBlaster AWE 32 (PnP), - SoundBlaster AWE 64 PnP - - mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) - csp - ASP/CSP chip support - 0 = disable (default), 1 = enable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for SB DSP 4.x chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) - (snd-sbawe module only) - irq - IRQ # for SB DSP 4.x chip (5,7,9,10) - dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) - dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) - - This module supports multiple cards, autoprobe and ISA PnP. - - Note: To use Vibra16X cards in 16-bit half duplex mode, you must - disable 16bit DMA with dma16 = -1 module parameter. - Also, all Sound Blaster 16 type cards can operate in 16-bit - half duplex mode through 8-bit DMA channel by disabling their - 16-bit DMA channel. - - The power-management is supported. - - Module snd-sc6000 - ----------------- - - Module for Gallant SC-6000 soundcard and later models: SC-6600 - and SC-7000. - - port - Port # (0x220 or 0x240) - mss_port - MSS Port # (0x530 or 0xe80) - irq - IRQ # (5,7,9,10,11) - mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq - dma - DMA # (1,3,0) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - - Module snd-sscape - ----------------- - - Module for ENSONIQ SoundScape cards. - - port - Port # (PnP setup) - wss_port - WSS Port # (PnP setup) - irq - IRQ # (PnP setup) - mpu_irq - MPU-401 IRQ # (PnP setup) - dma - DMA # (PnP setup) - dma2 - 2nd DMA # (PnP setup, -1 to disable) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - The driver requires the firmware loader support on kernel. - - Module snd-sun-amd7930 (on sparc only) - -------------------------------------- - - Module for AMD7930 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-cs4231 (on sparc only) - ------------------------------------- - - Module for CS4231 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-dbri (on sparc only) - ----------------------------------- - - Module for DBRI sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-wavefront - -------------------- - - Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. - - use_cs4232_midi - Use CS4232 MPU-401 interface - (inaccessibly located inside your computer) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - cs4232_pcm_port - Port # for CS4232 PCM interface. - cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). - cs4232_mpu_port - Port # for CS4232 MPU-401 interface. - cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). - ics2115_port - Port # for ICS2115 - ics2115_irq - IRQ # for ICS2115 - fm_port - FM OPL-3 Port # - dma1 - DMA1 # for CS4232 PCM interface. - dma2 - DMA2 # for CS4232 PCM interface. - - The below are options for wavefront_synth features: - wf_raw - Assume that we need to boot the OS (default:no) - If yes, then during driver loading, the state of the board is - ignored, and we reset the board and load the firmware anyway. - fx_raw - Assume that the FX process needs help (default:yes) - If false, we'll leave the FX processor in whatever state it is - when the driver is loaded. The default is to download the - microprogram and associated coefficients to set it up for - "default" operation, whatever that means. - debug_default - Debug parameters for card initialization - wait_usecs - How long to wait without sleeping, usecs - (default:150) - This magic number seems to give pretty optimal throughput - based on my limited experimentation. - If you want to play around with it and find a better value, be - my guest. Remember, the idea is to get a number that causes us - to just busy wait for as many WaveFront commands as possible, - without coming up with a number so large that we hog the whole - CPU. - Specifically, with this number, out of about 134,000 status - waits, only about 250 result in a sleep. - sleep_interval - How long to sleep when waiting for reply - (default: 100) - sleep_tries - How many times to try sleeping during a wait - (default: 50) - ospath - Pathname to processed ICS2115 OS firmware - (default:wavefront.os) - The path name of the ISC2115 OS firmware. In the recent - version, it's handled via firmware loader framework, so it - must be installed in the proper path, typically, - /lib/firmware. - reset_time - How long to wait for a reset to take effect - (default:2) - ramcheck_time - How many seconds to wait for the RAM test - (default:20) - osrun_time - How many seconds to wait for the ICS2115 OS - (default:10) - - This module supports multiple cards and ISA PnP. - - Note: the firmware file "wavefront.os" was located in the earlier - version in /etc. Now it's loaded via firmware loader, and - must be in the proper firmware path, such as /lib/firmware. - Copy (or symlink) the file appropriately if you get an error - regarding firmware downloading after upgrading the kernel. - - Module snd-sonicvibes - --------------------- - - Module for S3 SonicVibes PCI sound cards. - * PINE Schubert 32 PCI - - reverb - Reverb Enable - 1 = enable, 0 = disable (default) - - SoundCard must have onboard SRAM for this. - mge - Mic Gain Enable - 1 = enable, 0 = disable (default) - - This module supports multiple cards and autoprobe. - - Module snd-serial-u16550 - ------------------------ - - Module for UART16550A serial MIDI ports. - - port - port # for UART16550A chip - irq - IRQ # for UART16550A chip, -1 = poll mode - speed - speed in bauds (9600,19200,38400,57600,115200) - 38400 = default - base - base for divisor in bauds (57600,115200,230400,460800) - 115200 = default - outs - number of MIDI ports in a serial port (1-4) - 1 = default - adaptor - Type of adaptor. - 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, - 3 = MS-124W M/B, 4 = Generic - - This module supports multiple cards. This module does not support autoprobe - thus the main port must be specified!!! Other options are optional. - - Module snd-trident - ------------------ - - Module for Trident 4DWave DX/NX sound cards. - * Best Union Miss Melody 4DWave PCI - * HIS 4DWave PCI - * Warpspeed ONSpeed 4DWave PCI - * AzTech PCI 64-Q3D - * Addonics SV 750 - * CHIC True Sound 4Dwave - * Shark Predator4D-PCI - * Jaton SonicWave 4D - * SiS SI7018 PCI Audio - * Hoontech SoundTrack Digital 4DWave NX - - pcm_channels - max channels (voices) reserved for PCM - wavetable_size - max wavetable size in kB (4-?kb) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-ua101 - ---------------- - - Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. - - This module supports multiple devices, autoprobe and hotplugging. - - Module snd-usb-audio - -------------------- - - Module for USB audio and USB MIDI devices. - - vid - Vendor ID for the device (optional) - pid - Product ID for the device (optional) - nrpacks - Max. number of packets per URB (default: 8) - 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.d/alsa.conf - - install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader - - (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to - /etc/modules.conf, instead.) - IBL size defines the interrupts period for PCM. The smaller size - gives smaller latency but leads to more CPU consumption, too. - The size is usually aligned to 126. As default (=0), the smallest - size is chosen. The possible IBL values can be found in - /proc/asound/cardX/vx-status proc file. - - The power-management is supported. - - Module snd-vxpocket - ------------------- - - Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. - - ibl - Capture IBL size. (default = 0, minimum size) - - This module supports multiple cards. The module is compiled only when - PCMCIA is supported on kernel. - - With the older 2.6.x kernel, to activate the driver via the card - manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the - sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no - longer require a config file. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via vxloader utility in alsa-tools package. - - About capture IBL, see the description of snd-vx222 module. - - Note: snd-vxp440 driver is merged to snd-vxpocket driver since - ALSA 1.0.10. - - The power-management is supported. - - Module snd-ymfpci - ----------------- - - Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). - - mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default, - 1 (auto-detect for YMF744/754 only) - fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default - 1 (auto-detect for YMF744/754 only) - joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default, - 1 (auto-detect) - rear_switch - enable shared rear/line-in switch (bool) - - This module supports autoprobe and multiple chips. - - The power-management is supported. - - Module snd-pdaudiocf - -------------------- - - Module for Sound Core PDAudioCF sound card. - - The power-management is supported. - - -AC97 Quirk Option -================= - -The ac97_quirk option is used to enable/override the workaround for -specific devices on drivers for on-board AC'97 controllers like -snd-intel8x0. Some hardware have swapped output pins between Master -and Headphone, or Surround (thanks to confusion of AC'97 -specifications from version to version :-) - -The driver provides the auto-detection of known problematic devices, -but some might be unknown or wrongly detected. In such a case, pass -the proper value with this option. - -The following strings are accepted: - - default Don't override the default setting - - none Disable the quirk - - hp_only Bind Master and Headphone controls as a single control - - swap_hp Swap headphone and master controls - - swap_surround Swap master and surround controls - - ad_sharing For AD1985, turn on OMS bit and use headphone - - alc_jack For ALC65x, turn on the jack sense mode - - inv_eapd Inverted EAPD implementation - - mute_led Bind EAPD bit for turning on/off mute LED - -For backward compatibility, the corresponding integer value -1, 0, -... are accepted, too. - -For example, if "Master" volume control has no effect on your device -but only "Headphone" does, pass ac97_quirk=hp_only module option. - - -Configuring Non-ISAPNP Cards -============================ - -When the kernel is configured with ISA-PnP support, the modules -supporting the isapnp cards will have module options "isapnp". -If this option is set, *only* the ISA-PnP devices will be probed. -For probing the non ISA-PnP cards, you have to pass "isapnp=0" option -together with the proper i/o and irq configuration. - -When the kernel is configured without ISA-PnP support, isapnp option -will be not built in. - - -Module Autoloading Support -========================== - -The ALSA drivers can be loaded automatically on demand by defining -module aliases. The string 'snd-card-%1' is requested for ALSA native -devices where %i is sound card number from zero to seven. - -To auto-load an ALSA driver for OSS services, define the string -'sound-slot-%i' where %i means the slot number for OSS, which -corresponds to the card index of ALSA. Usually, define this -as the same card module. - -An example configuration for a single emu10k1 card is like below: ------ /etc/modprobe.d/alsa.conf -alias snd-card-0 snd-emu10k1 -alias sound-slot-0 snd-emu10k1 ------ /etc/modprobe.d/alsa.conf - -The available number of auto-loaded sound cards depends on the module -option "cards_limit" of snd module. As default it's set to 1. -To enable the auto-loading of multiple cards, specify the number of -sound cards in that option. - -When multiple cards are available, it'd better to specify the index -number for each card via module option, too, so that the order of -cards is kept consistent. - -An example configuration for two sound cards is like below: - ------ /etc/modprobe.d/alsa.conf -# ALSA portion -options snd cards_limit=2 -alias snd-card-0 snd-interwave -alias snd-card-1 snd-ens1371 -options snd-interwave index=0 -options snd-ens1371 index=1 -# OSS/Free portion -alias sound-slot-0 snd-interwave -alias sound-slot-1 snd-ens1371 ------ /etc/modprobe.d/alsa.conf - -In this example, the interwave card is always loaded as the first card -(index 0) and ens1371 as the second (index 1). - -Alternative (and new) way to fixate the slot assignment is to use -"slots" option of snd module. In the case above, specify like the -following: - -options snd slots=snd-interwave,snd-ens1371 - -Then, the first slot (#0) is reserved for snd-interwave driver, and -the second (#1) for snd-ens1371. You can omit index option in each -driver if slots option is used (although you can still have them at -the same time as long as they don't conflict). - -The slots option is especially useful for avoiding the possible -hot-plugging and the resultant slot conflict. For example, in the -case above again, the first two slots are already reserved. If any -other driver (e.g. snd-usb-audio) is loaded before snd-interwave or -snd-ens1371, it will be assigned to the third or later slot. - -When a module name is given with '!', the slot will be given for any -modules but that name. For example, "slots=!snd-pcsp" will reserve -the first slot for any modules but snd-pcsp. - - -ALSA PCM devices to OSS devices mapping -======================================= - -/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 -/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 -/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 -/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 -/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 -/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 -/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 -/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 -/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 - -The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means -sound card number and second means device number. The ALSA devices -have either 'c' or 'p' suffix indicating the direction, capture and -playback, respectively. - -Please note that the device mapping above may be varied via the module -options of snd-pcm-oss module. - - -Proc interfaces (/proc/asound) -============================== - -/proc/asound/card#/pcm#[cp]/oss -------------------------------- - String "erase" - erase all additional information about OSS applications - String "<app_name> <fragments> <fragment_size> [<options>]" - - <app_name> - name of application with (higher priority) or without path - <fragments> - number of fragments or zero if auto - <fragment_size> - size of fragment in bytes or zero if auto - <options> - optional parameters - - disable the application tries to open a pcm device for - this channel but does not want to use it. - (Cause a bug or mmap needs) - It's good for Quake etc... - - direct don't use plugins - - block force block mode (rvplayer) - - non-block force non-block mode - - whole-frag write only whole fragments (optimization affecting - playback only) - - no-silence do not fill silence ahead to avoid clicks - - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl - instead of filled blocks - - Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss - echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss - echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss - - -Early Buffer Allocation -======================= - -Some drivers (e.g. hdsp) require the large contiguous buffers, and -sometimes it's too late to find such spaces when the driver module is -actually loaded due to memory fragmentation. You can pre-allocate the -PCM buffers by loading snd-page-alloc module and write commands to its -proc file in prior, for example, in the early boot stage like -/etc/init.d/*.local scripts. - -Reading the proc file /proc/drivers/snd-page-alloc shows the current -usage of page allocation. In writing, you can send the following -commands to the snd-page-alloc driver: - - - add VENDOR DEVICE MASK SIZE BUFFERS - - VENDOR and DEVICE are PCI vendor and device IDs. They take - integer numbers (0x prefix is needed for the hex). - MASK is the PCI DMA mask. Pass 0 if not restricted. - SIZE is the size of each buffer to allocate. You can pass - k and m suffix for KB and MB. The max number is 16MB. - BUFFERS is the number of buffers to allocate. It must be greater - than 0. The max number is 4. - - - erase - - This will erase the all pre-allocated buffers which are not in - use. - - -Links and Addresses -=================== - - ALSA project homepage - http://www.alsa-project.org - - Kernel Bugzilla - http://bugzilla.kernel.org/ - - ALSA Developers ML - mailto:alsa-devel@alsa-project.org - - alsa-info.sh script - http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt deleted file mode 100644 index 7f10dc6ff28..00000000000 --- a/Documentation/sound/alsa/Audigy-mixer.txt +++ /dev/null @@ -1,345 +0,0 @@ - - Sound Blaster Audigy mixer / default DSP code - =========================================== - -This is based on SB-Live-mixer.txt. - -The EMU10K2 chips have a DSP part which can be programmed to support -various ways of sample processing, which is described here. -(This article does not deal with the overall functionality of the -EMU10K2 chips. See the manuals section for further details.) - -The ALSA driver programs this portion of chip by default code -(can be altered later) which offers the following functionality: - - -1) Digital mixer controls -------------------------- - -These controls are built using the DSP instructions. They offer extended -functionality. Only the default build-in code in the ALSA driver is described -here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). - - -Explanation of used abbreviations: - -DAC - digital to analog converter -ADC - analog to digital converter -I2S - one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) -LFE - low frequency effects (subwoofer signal) -AC97 - a chip containing an analog mixer, DAC and ADC converters -IEC958 - S/PDIF -FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators. - Each of the synthesizer voices can feed its output to these accumulators - and the DSP microcontroller can operate with the resulting sum. - -name='PCM Front Playback Volume',index=0 - -This control is used to attenuate samples for left and right front PCM FX-bus -accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM -samples for 5.1 playback. The result samples are forwarded to the front DAC PCM -slots of the Philips DAC. - -name='PCM Surround Playback Volume',index=0 - -This control is used to attenuate samples for left and right surround PCM FX-bus -accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM -samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM -slots of the Philips DAC. - -name='PCM Center Playback Volume',index=0 - -This control is used to attenuate samples for center PCM FX-bus accumulator. -ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample -is forwarded to the center DAC PCM slot of the Philips DAC. - -name='PCM LFE Playback Volume',index=0 - -This control is used to attenuate sample for LFE PCM FX-bus accumulator. -ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample -is forwarded to the LFE DAC PCM slot of the Philips DAC. - -name='PCM Playback Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for -stereo playback. The result samples are forwarded to the front DAC PCM slots -of the Philips DAC. - -name='PCM Capture Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Music Playback Volume',index=0 - -This control is used to attenuate samples for left and right MIDI FX-bus -accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. -The result samples are forwarded to the front DAC PCM slots of the AC97 codec. - -name='Music Capture Volume',index=0 - -These controls are used to attenuate samples for left and right MIDI FX-bus -accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Mic Playback Volume',index=0 - -This control is used to attenuate samples for left and right Mic input. -For Mic input is used AC97 codec. The result samples are forwarded to -the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic -capture FIFO (device 1 - 16bit/8KHz mono) too without volume control. - -name='Mic Capture Volume',index=0 - -This control is used to attenuate samples for left and right Mic input. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Audigy CD Playback Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 TTL -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the front DAC PCM slots of the Philips DAC. - -name='Audigy CD Capture Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 TTL -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the ADC capture FIFO (thus to the standard capture PCM device). - -name='IEC958 Optical Playback Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 optical -digital input. The result samples are forwarded to the front DAC PCM slots -of the Philips DAC. - -name='IEC958 Optical Capture Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 optical -digital inputs. The result samples are forwarded to the ADC capture FIFO -(thus to the standard capture PCM device). - -name='Line2 Playback Volume',index=0 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the AudigyDrive). The result samples are forwarded to the front -DAC PCM slots of the Philips DAC. - -name='Line2 Capture Volume',index=1 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the AudigyDrive). The result samples are forwarded to the ADC -capture FIFO (thus to the standard capture PCM device). - -name='Analog Mix Playback Volume',index=0 - -This control is used to attenuate samples from left and right I2S ADC -inputs from Philips ADC. The result samples are forwarded to the front -DAC PCM slots of the Philips DAC. This contains mix from analog sources -like CD, Line In, Aux, .... - -name='Analog Mix Capture Volume',index=1 - -This control is used to attenuate samples from left and right I2S ADC -inputs Philips ADC. The result samples are forwarded to the ADC -capture FIFO (thus to the standard capture PCM device). - -name='Aux2 Playback Volume',index=0 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the AudigyDrive). The result samples are forwarded to the front -DAC PCM slots of the Philips DAC. - -name='Aux2 Capture Volume',index=1 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the AudigyDrive). The result samples are forwarded to the ADC -capture FIFO (thus to the standard capture PCM device). - -name='Front Playback Volume',index=0 - -All stereo signals are mixed together and mirrored to surround, center and LFE. -This control is used to attenuate samples for left and right front speakers of -this mix. - -name='Surround Playback Volume',index=0 - -All stereo signals are mixed together and mirrored to surround, center and LFE. -This control is used to attenuate samples for left and right surround speakers of -this mix. - -name='Center Playback Volume',index=0 - -All stereo signals are mixed together and mirrored to surround, center and LFE. -This control is used to attenuate sample for center speaker of this mix. - -name='LFE Playback Volume',index=0 - -All stereo signals are mixed together and mirrored to surround, center and LFE. -This control is used to attenuate sample for LFE speaker of this mix. - -name='Tone Control - Switch',index=0 - -This control turns the tone control on or off. The samples for front, rear -and center / LFE outputs are affected. - -name='Tone Control - Bass',index=0 - -This control sets the bass intensity. There is no neutral value!! -When the tone control code is activated, the samples are always modified. -The closest value to pure signal is 20. - -name='Tone Control - Treble',index=0 - -This control sets the treble intensity. There is no neutral value!! -When the tone control code is activated, the samples are always modified. -The closest value to pure signal is 20. - -name='Master Playback Volume',index=0 - -This control is used to attenuate samples for front, surround, center and -LFE outputs. - -name='IEC958 Optical Raw Playback Switch',index=0 - -If this switch is on, then the samples for the IEC958 (S/PDIF) digital -output are taken only from the raw FX8010 PCM, otherwise standard front -PCM samples are taken. - - -2) PCM stream related controls ------------------------------- - -name='EMU10K1 PCM Volume',index 0-31 - -Channel volume attenuation in range 0-0xffff. The maximum value (no -attenuation) is default. The channel mapping for three values is -as follows: - - 0 - mono, default 0xffff (no attenuation) - 1 - left, default 0xffff (no attenuation) - 2 - right, default 0xffff (no attenuation) - -name='EMU10K1 PCM Send Routing',index 0-31 - -This control specifies the destination - FX-bus accumulators. There 24 -values with this mapping: - - 0 - mono, A destination (FX-bus 0-63), default 0 - 1 - mono, B destination (FX-bus 0-63), default 1 - 2 - mono, C destination (FX-bus 0-63), default 2 - 3 - mono, D destination (FX-bus 0-63), default 3 - 4 - mono, E destination (FX-bus 0-63), default 0 - 5 - mono, F destination (FX-bus 0-63), default 0 - 6 - mono, G destination (FX-bus 0-63), default 0 - 7 - mono, H destination (FX-bus 0-63), default 0 - 8 - left, A destination (FX-bus 0-63), default 0 - 9 - left, B destination (FX-bus 0-63), default 1 - 10 - left, C destination (FX-bus 0-63), default 2 - 11 - left, D destination (FX-bus 0-63), default 3 - 12 - left, E destination (FX-bus 0-63), default 0 - 13 - left, F destination (FX-bus 0-63), default 0 - 14 - left, G destination (FX-bus 0-63), default 0 - 15 - left, H destination (FX-bus 0-63), default 0 - 16 - right, A destination (FX-bus 0-63), default 0 - 17 - right, B destination (FX-bus 0-63), default 1 - 18 - right, C destination (FX-bus 0-63), default 2 - 19 - right, D destination (FX-bus 0-63), default 3 - 20 - right, E destination (FX-bus 0-63), default 0 - 21 - right, F destination (FX-bus 0-63), default 0 - 22 - right, G destination (FX-bus 0-63), default 0 - 23 - right, H destination (FX-bus 0-63), default 0 - -Don't forget that it's illegal to assign a channel to the same FX-bus accumulator -more than once (it means 0=0 && 1=0 is an invalid combination). - -name='EMU10K1 PCM Send Volume',index 0-31 - -It specifies the attenuation (amount) for given destination in range 0-255. -The channel mapping is following: - - 0 - mono, A destination attn, default 255 (no attenuation) - 1 - mono, B destination attn, default 255 (no attenuation) - 2 - mono, C destination attn, default 0 (mute) - 3 - mono, D destination attn, default 0 (mute) - 4 - mono, E destination attn, default 0 (mute) - 5 - mono, F destination attn, default 0 (mute) - 6 - mono, G destination attn, default 0 (mute) - 7 - mono, H destination attn, default 0 (mute) - 8 - left, A destination attn, default 255 (no attenuation) - 9 - left, B destination attn, default 0 (mute) - 10 - left, C destination attn, default 0 (mute) - 11 - left, D destination attn, default 0 (mute) - 12 - left, E destination attn, default 0 (mute) - 13 - left, F destination attn, default 0 (mute) - 14 - left, G destination attn, default 0 (mute) - 15 - left, H destination attn, default 0 (mute) - 16 - right, A destination attn, default 0 (mute) - 17 - right, B destination attn, default 255 (no attenuation) - 18 - right, C destination attn, default 0 (mute) - 19 - right, D destination attn, default 0 (mute) - 20 - right, E destination attn, default 0 (mute) - 21 - right, F destination attn, default 0 (mute) - 22 - right, G destination attn, default 0 (mute) - 23 - right, H destination attn, default 0 (mute) - - - -4) MANUALS/PATENTS: -------------------- - -ftp://opensource.creative.com/pub/doc -------------------------------------- - - Files: - LM4545.pdf AC97 Codec - - m2049.pdf The EMU10K1 Digital Audio Processor - - hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects - - -WIPO Patents ------------- - Patent numbers: - WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) - streams - - WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) - - WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction - Execution and Audio Data Sequencing (Jan. 14, 1999) - - -US Patents (http://www.uspto.gov/) ----------------------------------- - - US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) - - US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) - with a multiport memory onto which multiple asynchronous - digital sound samples can be concurrently loaded - - US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) - - US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) - - US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) - system bus with prioritization and modification of bus transfers - in accordance with loop ends and minimum block sizes - - US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) - pool of short term memory registers - - US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) - a common interrupt by associating programs to GP registers, - defining interrupt register, polling GP registers, and invoking - callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt deleted file mode 100644 index 654dd3b694a..00000000000 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ /dev/null @@ -1,442 +0,0 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5 - ======================================================== - - Thibault Le Meur <Thibault.LeMeur@supelec.fr> - -This document is a guide to using the M-Audio Audiophile USB (tm) device with -ALSA and JACK. - -History -======= -* v1.4 - Thibault Le Meur (2007-07-11) - - Added Low Endianness nature of 16bits-modes - found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se> - - Modifying document structure -* v1.5 - Thibault Le Meur (2007-07-12) - - Added AC3/DTS passthru info - - -1 - Audiophile USB Specs and correct usage -========================================== - -This part is a reminder of important facts about the functions and limitations -of the device. - -The device has 4 audio interfaces, and 2 MIDI ports: - * Analog Stereo Input (Ai) - - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) - - When the 1/4" TS (jack) connectors are connected, the RCA connectors - are disabled - * Analog Stereo Output (Ao) - * Digital Stereo Input (Di) - * Digital Stereo Output (Do) - * Midi In (Mi) - * Midi Out (Mo) - -The internal DAC/ADC has the following characteristics: -* sample depth of 16 or 24 bits -* sample rate from 8kHz to 96kHz -* Two interfaces can't use different sample depths at the same time. -Moreover, the Audiophile USB documentation gives the following Warning: -"Please exit any audio application running before switching between bit depths" - -Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be -activated at the same time depending on the audio mode selected: - * 16-bit/48kHz ==> 4 channels in + 4 channels out - - Ai+Ao+Di+Do - * 24-bit/48kHz ==> 4 channels in + 2 channels out, - or 2 channels in + 4 channels out - - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do - * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only) - - Ai or Ao or Di or Do - -Important facts about the Digital interface: --------------------------------------------- - * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, -though I haven't tested it under Linux - - Note that in this setup only the Do interface can be enabled - * Apart from recording an audio digital stream, enabling the Di port is a way -to synchronize the device to an external sample clock - - As a consequence, the Di port must be enable only if an active Digital -source is connected - - Enabling Di when no digital source is connected can result in a -synchronization error (for instance sound played at an odd sample rate) - - -2 - Audiophile USB MIDI support in ALSA -======================================= - -The Audiophile USB MIDI ports will be automatically supported once the -following modules have been loaded: - * snd-usb-audio - * snd-seq-midi - -No additional setting is required. - - -3 - Audiophile USB Audio support in ALSA -======================================== - -Audio functions of the Audiophile USB device are handled by the snd-usb-audio -module. This module can work in a default mode (without any device-specific -parameter), or in an "advanced" mode with the device-specific parameter called -"device_setup". - -3.1 - Default Alsa driver mode ------------------------------- - -The default behavior of the snd-usb-audio driver is to list the device -capabilities at startup and activate the required mode when required -by the applications: for instance if the user is recording in a -24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode, -the snd-usb-audio module will reconfigure the device on the fly. - -This approach has the advantage to let the driver automatically switch from sample -rates/depths automatically according to the user's needs. However, those who -are using the device under windows know that this is not how the device is meant to -work: under windows applications must be closed before using the m-audio control -panel to switch the device working mode. Thus as we'll see in next section, this -Default Alsa driver mode can lead to device misconfigurations. - -Let's get back to the Default Alsa driver mode for now. In this case the -Audiophile interfaces are mapped to alsa pcm devices in the following -way (I suppose the device's index is 1): - * hw:1,0 is Ao in playback and Di in capture - * hw:1,1 is Do in playback and Ai in capture - * hw:1,2 is Do in AC3/DTS passthrough mode - -In this mode, the device uses Big Endian byte-encoding so that -supported audio format are S16_BE for 16-bit depth modes and S24_3BE for -24-bits depth mode. - -One exception is the hw:1,2 port which was reported to be Little Endian -compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams. -This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface -is reported to be big endian in this default driver mode. - -Examples: - * playing a S24_3BE encoded raw file to the Ao port - % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw - * recording a S24_3BE encoded raw file from the Ai port - % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw - * playing a S16_BE encoded raw file to the Do port - % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw - * playing an ac3 sample file to the Do port - % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw - -If you're happy with the default Alsa driver mode and don't experience any -issue with this mode, then you can skip the following chapter. - -3.2 - Advanced module setup ---------------------------- - -Due to the hardware constraints described above, the device initialization made -by the Alsa driver in default mode may result in a corrupted state of the -device. For instance, a particularly annoying issue is that the sound captured -from the Ai interface sounds distorted (as if boosted with an excessive high -volume gain). - -For people having this problem, the snd-usb-audio module has a new module -parameter called "device_setup" (this parameter was introduced in kernel -release 2.6.17) - -3.2.1 - Initializing the working mode of the Audiophile USB - -As far as the Audiophile USB device is concerned, this value let the user -specify: - * the sample depth - * the sample rate - * whether the Di port is used or not - -When initialized with "device_setup=0x00", the snd-usb-audio module has -the same behaviour as when the parameter is omitted (see paragraph "Default -Alsa driver mode" above) - -Others modes are described in the following subsections. - -3.2.1.1 - 16-bit modes - -The two supported modes are: - - * device_setup=0x01 - - 16bits 48kHz mode with Di disabled - - Ai,Ao,Do can be used at the same time - - hw:1,0 is not available in capture mode - - hw:1,2 is not available - - * device_setup=0x11 - - 16bits 48kHz mode with Di enabled - - Ai,Ao,Di,Do can be used at the same time - - hw:1,0 is available in capture mode - - hw:1,2 is not available - -In this modes the device operates only at 16bits-modes. Before kernel 2.6.23, -the devices where reported to be Big-Endian when in fact they were Little-Endian -so that playing a file was a matter of using: - % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw -where "test_S16_LE.raw" was in fact a little-endian sample file. - -Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in -these modes) a fix has been committed (expected in kernel 2.6.23) and -Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as -using: - % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw - -3.2.1.2 - 24-bit modes - -The three supported modes are: - - * device_setup=0x09 - - 24bits 48kHz mode with Di disabled - - Ai,Ao,Do can be used at the same time - - hw:1,0 is not available in capture mode - - hw:1,2 is not available - - * device_setup=0x19 - - 24bits 48kHz mode with Di enabled - - 3 ports from {Ai,Ao,Di,Do} can be used at the same time - - hw:1,0 is available in capture mode and an active digital source must be - connected to Di - - hw:1,2 is not available - - * device_setup=0x0D or 0x10 - - 24bits 96kHz mode - - Di is enabled by default for this mode but does not need to be connected - to an active source - - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time - - hw:1,0 is available in captured mode - - hw:1,2 is not available - -In these modes the device is only Big-Endian compliant (see "Default Alsa driver -mode" above for an aplay command example) - -3.2.1.3 - AC3 w/ DTS passthru mode - -Thanks to Hakan Lennestal, I now have a report saying that this mode works. - - * device_setup=0x03 - - 16bits 48kHz mode with only the Do port enabled - - AC3 with DTS passthru - - Caution with this setup the Do port is mapped to the pcm device hw:1,0 - -The command line used to playback the AC3/DTS encoded .wav-files in this mode: - % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw - -3.2.2 - How to use the device_setup parameter ----------------------------------------------- - -The parameter can be given: - - * By manually probing the device (as root): - # modprobe -r snd-usb-audio - # modprobe snd-usb-audio index=1 device_setup=0x09 - - * Or while configuring the modules options in your modules configuration file - (tipically a .conf file in /etc/modprobe.d/ directory: - alias snd-card-1 snd-usb-audio - options snd-usb-audio index=1 device_setup=0x09 - -CAUTION when initializing the device -------------------------------------- - - * Correct initialization on the device requires that device_setup is given to - the module BEFORE the device is turned on. So, if you use the "manual probing" - method described above, take care to power-on the device AFTER this initialization. - - * Failing to respect this will lead to a misconfiguration of the device. In this case - turn off the device, unprobe the snd-usb-audio module, then probe it again with - correct device_setup parameter and then (and only then) turn on the device again. - - * If you've correctly initialized the device in a valid mode and then want to switch - to another mode (possibly with another sample-depth), please use also the following - procedure: - - first turn off the device - - de-register the snd-usb-audio module (modprobe -r) - - change the device_setup parameter by changing the device_setup - option in /etc/modprobe.d/*.conf - - turn on the device - * A workaround for this last issue has been applied to kernel 2.6.23, but it may not - be enough to ensure the 'stability' of the device initialization. - -3.2.3 - Technical details for hackers -------------------------------------- -This section is for hackers, wanting to understand details about the device -internals and how Alsa supports it. - -3.2.3.1 - Audiophile USB's device_setup structure - -If you want to understand the device_setup magic numbers for the Audiophile -USB, you need some very basic understanding of binary computation. However, -this is not required to use the parameter and you may skip this section. - -The device_setup is one byte long and its structure is the following: - - +---+---+---+---+---+---+---+---+ - | b7| b6| b5| b4| b3| b2| b1| b0| - +---+---+---+---+---+---+---+---+ - | 0 | 0 | 0 | Di|24B|96K|DTS|SET| - +---+---+---+---+---+---+---+---+ - -Where: - * b0 is the "SET" bit - - it MUST be set if device_setup is initialized - * b1 is the "DTS" bit - - it is set only for Digital output with DTS/AC3 - - this setup is not tested - * b2 is the Rate selection flag - - When set to "1" the rate range is 48.1-96kHz - - Otherwise the sample rate range is 8-48kHz - * b3 is the bit depth selection flag - - When set to "1" samples are 24bits long - - Otherwise they are 16bits long - - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits - samples - * b4 is the Digital input flag - - When set to "1" the device assumes that an active digital source is - connected - - You shouldn't enable Di if no source is seen on the port (this leads to - synchronization issues) - - b4 is implied by b2 (since only one port is enabled at a time no synch - error can occur) - * b5 to b7 are reserved for future uses, and must be set to "0" - - might become Ao, Do, Ai, for b7, b6, b4 respectively - -Caution: - * there is no check on the value you will give to device_setup - - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since - b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages - * Hardware constraints due to the USB bus limitation aren't checked - - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll - only be able to use one at the same time - -3.2.3.2 - USB implementation details for this device - -You may safely skip this section if you're not interested in driver -hacking. - -This section describes some internal aspects of the device and summarizes the -data I got by usb-snooping the windows and Linux drivers. - -The M-Audio Audiophile USB has 7 USB Interfaces: -a "USB interface": - * USB Interface nb.0 - * USB Interface nb.1 - - Audio Control function - * USB Interface nb.2 - - Analog Output - * USB Interface nb.3 - - Digital Output - * USB Interface nb.4 - - Analog Input - * USB Interface nb.5 - - Digital Input - * USB Interface nb.6 - - MIDI interface compliant with the MIDIMAN quirk - -Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: - * Interface 3 (Digital Out) has an extra Alset nb.6 - * Interface 5 (Digital In) does not have Alset nb.3 and 5 - -Here is a short description of the AltSettings capabilities: - * AltSettings 1 corresponds to - - 24-bit depth, 48.1-96kHz sample mode - - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) - * AltSettings 2 corresponds to - - 24-bit depth, 8-48kHz sample mode - - Asynch capture and playback (Ao,Ai,Do,Di) - * AltSettings 3 corresponds to - - 24-bit depth, 8-48kHz sample mode - - Synch capture (Ai) and Adaptive playback (Ao,Do) - * AltSettings 4 corresponds to - - 16-bit depth, 8-48kHz sample mode - - Asynch capture and playback (Ao,Ai,Do,Di) - * AltSettings 5 corresponds to - - 16-bit depth, 8-48kHz sample mode - - Synch capture (Ai) and Adaptive playback (Ao,Do) - * AltSettings 6 corresponds to - - 16-bit depth, 8-48kHz sample mode - - Synch playback (Do), audio format type III IEC1937_AC-3 - -In order to ensure a correct initialization of the device, the driver -_must_know_ how the device will be used: - * if DTS is chosen, only Interface 2 with AltSet nb.6 must be - registered - * if 96KHz only AltSets nb.1 of each interface must be selected - * if samples are using 24bits/48KHz then AltSet 2 must me used if - Digital input is connected, and only AltSet nb.3 if Digital input - is not connected - * if samples are using 16bits/48KHz then AltSet 4 must me used if - Digital input is connected, and only AltSet nb.5 if Digital input - is not connected - -When device_setup is given as a parameter to the snd-usb-audio module, the -parse_audio_endpoints function uses a quirk called -"audiophile_skip_setting_quirk" in order to prevent AltSettings not -corresponding to device_setup from being registered in the driver. - -4 - Audiophile USB and Jack support -=================================== - -This section deals with support of the Audiophile USB device in Jack. - -There are 2 main potential issues when using Jackd with the device: -* support for Big-Endian devices in 24-bit modes -* support for 4-in / 4-out channels - -4.1 - Direct support in Jackd ------------------------------ - -Jack supports big endian devices only in recent versions (thanks to -Andreas Steinmetz for his first big-endian patch). I can't remember -exactly when this support was released into jackd, let's just say that -with jackd version 0.103.0 it's almost ok (just a small bug is affecting -16bits Big-Endian devices, but since you've read carefully the above -paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices -are now Little Endians ;-) ). - -You can run jackd with the following command for playback with Ao and -record with Ai: - % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 - -4.2 - Using Alsa plughw ------------------------ -If you don't have a recent Jackd installed, you can downgrade to using -the Alsa "plug" converter. - -For instance here is one way to run Jack with 2 playback channels on Ao and 2 -capture channels from Ai: - % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 - -However you may see the following warning message: -"You appear to be using the ALSA software "plug" layer, probably a result of -using the "default" ALSA device. This is less efficient than it could be. -Consider using a hardware device instead rather than using the plug layer." - -4.3 - Getting 2 input and/or output interfaces in Jack ------------------------------------------------------- - -As you can see, starting the Jack server this way will only enable 1 stereo -input (Di or Ai) and 1 stereo output (Ao or Do). - -This is due to the following restrictions: -* Jack can only open one capture device and one playback device at a time -* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1 - (and optionally hw:1,2) - -If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to -combine the Alsa devices into one logical "complex" device. - -If you want to give it a try, I recommend reading the information from -this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html -It is related to another device (ice1712) but can be adapted to suit -the Audiophile USB. - -Enabling multiple Audiophile USB interfaces for Jackd will certainly require: -* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page) -* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page) -* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc - file -* start jackd with this device - -I had no success in testing this for now, if you have any success with this kind -of setup, please drop me an email. diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt deleted file mode 100644 index f158cde8b06..00000000000 --- a/Documentation/sound/alsa/Bt87x.txt +++ /dev/null @@ -1,78 +0,0 @@ -Intro -===== - -You might have noticed that the bt878 grabber cards have actually -_two_ PCI functions: - -$ lspci -[ ... ] -00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02) -00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02) -[ ... ] - -The first does video, it is backward compatible to the bt848. The second -does audio. snd-bt87x is a driver for the second function. It's a sound -driver which can be used for recording sound (and _only_ recording, no -playback). As most TV cards come with a short cable which can be plugged -into your sound card's line-in you probably don't need this driver if all -you want to do is just watching TV... - -Some cards do not bother to connect anything to the audio input pins of -the chip, and some other cards use the audio function to transport MPEG -video data, so it's quite possible that audio recording may not work -with your card. - - -Driver Status -============= - -The driver is now stable. However, it doesn't know about many TV cards, -and it refuses to load for cards it doesn't know. - -If the driver complains ("Unknown TV card found, the audio driver will -not load"), you can specify the load_all=1 option to force the driver to -try to use the audio capture function of your card. If the frequency of -recorded data is not right, try to specify the digital_rate option with -other values than the default 32000 (often it's 44100 or 64000). - -If you have an unknown card, please mail the ID and board name to -<alsa-devel@alsa-project.org>, regardless of whether audio capture works -or not, so that future versions of this driver know about your card. - - -Audio modes -=========== - -The chip knows two different modes (digital/analog). snd-bt87x -registers two PCM devices, one for each mode. They cannot be used at -the same time. - - -Digital audio mode -================== - -The first device (hw:X,0) gives you 16 bit stereo sound. The sample -rate depends on the external source which feeds the Bt87x with digital -sound via I2S interface. - - -Analog audio mode (A/D) -======================= - -The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported -sample rates are between 119466 and 448000 Hz (yes, these numbers are -that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the -maximum sample rate is 1792000 Hz, but audio data becomes unusable -beyond 896000 Hz on my card. - -The chip has three analog inputs. Consequently you'll get a mixer -device to control these. - - -Have fun, - - Clemens - - -Written by Clemens Ladisch <clemens@ladisch.de> -big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org> diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt deleted file mode 100644 index 16935c8561f..00000000000 --- a/Documentation/sound/alsa/CMIPCI.txt +++ /dev/null @@ -1,254 +0,0 @@ - Brief Notes on C-Media 8338/8738/8768/8770 Driver - ================================================= - - Takashi Iwai <tiwai@suse.de> - - -Front/Rear Multi-channel Playback ---------------------------------- - -CM8x38 chip can use ADC as the second DAC so that two different stereo -channels can be used for front/rear playbacks. Since there are two -DACs, both streams are handled independently unlike the 4/6ch multi- -channel playbacks in the section below. - -As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for -card#0) for front and 4/6ch playbacks, while the second PCM device -(hw:0,1) is assigned to the second DAC for rear playback. - -There are slight differences between the two DACs: - -- The first DAC supports U8 and S16LE formats, while the second DAC - supports only S16LE. -- The second DAC supports only two channel stereo. - -Please note that the CM8x38 DAC doesn't support continuous playback -rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000, -44100 and 48000 Hz. - -The rear output can be heard only when "Four Channel Mode" switch is -disabled. Otherwise no signal will be routed to the rear speakers. -As default it's turned on. - -*** WARNING *** -When "Four Channel Mode" switch is off, the output from rear speakers -will be FULL VOLUME regardless of Master and PCM volumes. -This might damage your audio equipment. Please disconnect speakers -before your turn off this switch. -*** WARNING *** - -[ Well.. I once got the output with correct volume (i.e. same with the - front one) and was so excited. It was even with "Four Channel" bit - on and "double DAC" mode. Actually I could hear separate 4 channels - from front and rear speakers! But.. after reboot, all was gone. - It's a very pity that I didn't save the register dump at that - time.. Maybe there is an unknown register to achieve this... ] - -If your card has an extra output jack for the rear output, the rear -playback should be routed there as default. If not, there is a -control switch in the driver "Line-In As Rear", which you can change -via alsamixer or somewhat else. When this switch is on, line-in jack -is used as rear output. - -There are two more controls regarding to the rear output. -The "Exchange DAC" switch is used to exchange front and rear playback -routes, i.e. the 2nd DAC is output from front output. - - -4/6 Multi-Channel Playback --------------------------- - -The recent CM8738 chips support for the 4/6 multi-channel playback -function. This is useful especially for AC3 decoding. - -When the multi-channel is supported, the driver name has a suffix -"-MC" such like "CMI8738-MC6". You can check this name from -/proc/asound/cards. - -When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or -4) channels. While the dual DAC supports two different rates or -formats, the 4/6-ch playback supports only the same condition for all -channels. Since the multi-channel playback mode uses both DACs, you -cannot operate with full-duplex. - -The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51" -in alsa-lib. For example, you can play a WAV file with 6 channels like - - % aplay -Dsurround51 sixchannels.wav - -For programming the 4/6 channel playback, you need to specify the PCM -channels as you like and set the format S16LE. For example, for playback -with 4 channels, - - snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED); - // or mmap if you like - snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE); - snd_pcm_hw_params_set_channels(pcm, hw, 4); - -and use the interleaved 4 channel data. - -There are some control 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 deleted file mode 100644 index fea65bb6269..00000000000 --- a/Documentation/sound/alsa/ControlNames.txt +++ /dev/null @@ -1,85 +0,0 @@ -This document describes standard names of mixer controls. - -Syntax: SOURCE [DIRECTION] FUNCTION - -DIRECTION: - <nothing> (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-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt deleted file mode 100644 index e9621e349e1..00000000000 --- a/Documentation/sound/alsa/HD-Audio-Controls.txt +++ /dev/null @@ -1,116 +0,0 @@ -This file explains the codec-specific mixer controls. - -Realtek codecs --------------- - -* Channel Mode - This is an enum control to change the surround-channel setup, - appears only when the surround channels are available. - It gives the number of channels to be used, "2ch", "4ch", "6ch", - and "8ch". According to the configuration, this also controls the - jack-retasking of multi-I/O jacks. - -* Auto-Mute Mode - This is an enum control to change the auto-mute behavior of the - headphone and line-out jacks. If built-in speakers and headphone - and/or line-out jacks are available on a machine, this controls - appears. - When there are only either headphones or line-out jacks, it gives - "Disabled" and "Enabled" state. When enabled, the speaker is muted - automatically when a jack is plugged. - - When both headphone and line-out jacks are present, it gives - "Disabled", "Speaker Only" and "Line-Out+Speaker". When - speaker-only is chosen, plugging into a headphone or a line-out jack - mutes the speakers, but not line-outs. When line-out+speaker is - selected, plugging to a headphone jack mutes both speakers and - line-outs. - - -IDT/Sigmatel codecs -------------------- - -* Analog Loopback - This control enables/disables the analog-loopback circuit. This - appears only when "loopback" is set to true in a codec hint - (see HD-Audio.txt). Note that on some codecs the analog-loopback - and the normal PCM playback are exclusive, i.e. when this is on, you - won't hear any PCM stream. - -* Swap Center/LFE - Swaps the center and LFE channel order. Normally, the left - corresponds to the center and the right to the LFE. When this is - ON, the left to the LFE and the right to the center. - -* Headphone as Line Out - When this control is ON, treat the headphone jacks as line-out - jacks. That is, the headphone won't auto-mute the other line-outs, - and no HP-amp is set to the pins. - -* Mic Jack Mode, Line Jack Mode, etc - These enum controls the direction and the bias of the input jack - pins. Depending on the jack type, it can set as "Mic In" and "Line - In", for determining the input bias, or it can be set to "Line Out" - when the pin is a multi-I/O jack for surround channels. - - -VIA codecs ----------- - -* Smart 5.1 - An enum control to re-task the multi-I/O jacks for surround outputs. - When it's ON, the corresponding input jacks (usually a line-in and a - mic-in) are switched as the surround and the CLFE output jacks. - -* Independent HP - When this enum control is enabled, the headphone output is routed - from an individual stream (the third PCM such as hw:0,2) instead of - the primary stream. In the case the headphone DAC is shared with a - side or a CLFE-channel DAC, the DAC is switched to the headphone - automatically. - -* Loopback Mixing - An enum control to determine whether the analog-loopback route is - enabled or not. When it's enabled, the analog-loopback is mixed to - the front-channel. Also, the same route is used for the headphone - and speaker outputs. As a side-effect, when this mode is set, the - individual volume controls will be no longer available for - headphones and speakers because there is only one DAC connected to a - mixer widget. - -* Dynamic Power-Control - This control determines whether the dynamic power-control per jack - detection is enabled or not. When enabled, the widgets power state - (D0/D3) are changed dynamically depending on the jack plugging - state for saving power consumptions. However, if your system - doesn't provide a proper jack-detection, this won't work; in such a - case, turn this control OFF. - -* Jack Detect - This control is provided only for VT1708 codec which gives no proper - unsolicited event per jack plug. When this is on, the driver polls - the jack detection so that the headphone auto-mute can work, while - turning this off would reduce the power consumption. - - -Conexant codecs ---------------- - -* Auto-Mute Mode - See Reatek codecs. - - -Analog codecs --------------- - -* Channel Mode - This is an enum control to change the surround-channel setup, - appears only when the surround channels are available. - It gives the number of channels to be used, "2ch", "4ch" and "6ch". - According to the configuration, this also controls the - jack-retasking of multi-I/O jacks. - -* Independent HP - When this enum control is enabled, the headphone output is routed - from an individual stream (the third PCM such as hw:0,2) instead of - the primary stream. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt deleted file mode 100644 index 03f7897c641..00000000000 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ /dev/null @@ -1,284 +0,0 @@ - Model name Description - ---------- ----------- -ALC880 -====== - 3stack 3-jack in back and a headphone out - 3stack-digout 3-jack in back, a HP out and a SPDIF out - 5stack 5-jack in back, 2-jack in front - 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out - 6stack 6-jack in back, 2-jack in front - 6stack-digout 6-jack with a SPDIF out - -ALC260 -====== - N/A - -ALC262 -====== - N/A - -ALC267/268 -========== - N/A - -ALC269 -====== - laptop-amic Laptops with analog-mic input - laptop-dmic Laptops with digital-mic input - -ALC662/663/272 -============== - asus-mode1 ASUS - asus-mode2 ASUS - asus-mode3 ASUS - asus-mode4 ASUS - asus-mode5 ASUS - asus-mode6 ASUS - asus-mode7 ASUS - asus-mode8 ASUS - -ALC680 -====== - N/A - -ALC882/883/885/888/889 -====================== - acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G - acer-aspire-8930g Acer Aspire 8330G/6935G - acer-aspire Acer Aspire others - -ALC861/660 -========== - N/A - -ALC861VD/660VD -============== - N/A - -CMI9880 -======= - minimal 3-jack in back - min_fp 3-jack in back, 2-jack in front - full 6-jack in back, 2-jack in front - full_dig 6-jack in back, 2-jack in front, SPDIF I/O - allout 5-jack in back, 2-jack in front, SPDIF out - auto auto-config reading BIOS (default) - -AD1882 / AD1882A -================ - 3stack 3-stack mode (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 - 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 - dell-vostro-3500 Dell Vostro 3500 laptop - 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 deleted file mode 100644 index 7813c06a5c7..00000000000 --- a/Documentation/sound/alsa/HD-Audio.txt +++ /dev/null @@ -1,779 +0,0 @@ -MORE NOTES ON HD-AUDIO DRIVER -============================= - Takashi Iwai <tiwai@suse.de> - - -GENERAL -------- - -HD-audio is the new standard on-board audio component on modern PCs -after AC97. Although Linux has been supporting HD-audio since long -time ago, there are often problems with new machines. A part of the -problem is broken BIOS, and the rest is the driver implementation. -This document explains the brief trouble-shooting and debugging -methods for the HD-audio hardware. - -The HD-audio component consists of two parts: the controller chip and -the codec chips on the HD-audio bus. Linux provides a single driver -for all controllers, snd-hda-intel. Although the driver name contains -a word of a well-known hardware vendor, it's not specific to it but for -all controller chips by other companies. Since the HD-audio -controllers are supposed to be compatible, the single snd-hda-driver -should work in most cases. But, not surprisingly, there are known -bugs and issues specific to each controller type. The snd-hda-intel -driver has a bunch of workarounds for these as described below. - -A controller may have multiple codecs. Usually you have one audio -codec and optionally one modem codec. In theory, there might be -multiple audio codecs, e.g. for analog and digital outputs, and the -driver might not work properly because of conflict of mixer elements. -This should be fixed in future if such hardware really exists. - -The snd-hda-intel driver has several different codec parsers depending -on the codec. It has a generic parser as a fallback, but this -functionality is fairly limited until now. Instead of the generic -parser, usually the codec-specific parser (coded in patch_*.c) is used -for the codec-specific implementations. The details about the -codec-specific problems are explained in the later sections. - -If you are interested in the deep debugging of HD-audio, read the -HD-audio specification at first. The specification is found on -Intel's web page, for example: - -- http://www.intel.com/standards/hdaudio/ - - -HD-AUDIO CONTROLLER -------------------- - -DMA-Position Problem -~~~~~~~~~~~~~~~~~~~~ -The most common problem of the controller is the inaccurate DMA -pointer reporting. The DMA pointer for playback and capture can be -read in two ways, either via a LPIB register or via a position-buffer -map. As default the driver tries to read from the io-mapped -position-buffer, and falls back to LPIB if the position-buffer appears -dead. However, this detection isn't perfect on some devices. In such -a case, you can change the default method via `position_fix` option. - -`position_fix=1` means to use LPIB method explicitly. -`position_fix=2` means to use the position-buffer. -`position_fix=3` means to use a combination of both methods, needed -for some VIA controllers. The capture stream position is corrected -by comparing both LPIB and position-buffer values. -`position_fix=4` is another combination available for all controllers, -and uses LPIB for the playback and the position-buffer for the capture -streams. -0 is the default value for all other -controllers, the automatic check and fallback to LPIB as described in -the above. If you get a problem of repeated sounds, this option might -help. - -In addition to that, every controller is known to be broken regarding -the wake-up timing. It wakes up a few samples before actually -processing the data on the buffer. This caused a lot of problems, for -example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts -an artificial delay to the wake up timing. This delay is controlled -via `bdl_pos_adj` option. - -When `bdl_pos_adj` is a negative value (as default), it's assigned to -an appropriate value depending on the controller chip. For Intel -chips, it'd be 1 while it'd be 32 for others. Usually this works. -Only in case it doesn't work and you get warning messages, you should -change this parameter to other values. - - -Codec-Probing Problem -~~~~~~~~~~~~~~~~~~~~~ -A less often but a more severe problem is the codec probing. When -BIOS reports the available codec slots wrongly, the driver gets -confused and tries to access the non-existing codec slot. This often -results in the total screw-up, and destructs the further communication -with the codec chips. The symptom appears usually as error messages -like: ------------------------------------------------------------------------- - hda_intel: azx_get_response timeout, switching to polling mode: - last cmd=0x12345678 - hda_intel: azx_get_response timeout, switching to single_cmd mode: - last cmd=0x12345678 ------------------------------------------------------------------------- - -The first line is a warning, and this is usually relatively harmless. -It means that the codec response isn't notified via an IRQ. The -driver uses explicit polling method to read the response. It gives -very slight CPU overhead, but you'd unlikely notice it. - -The second line is, however, a fatal error. If this happens, usually -it means that something is really wrong. Most likely you are -accessing a non-existing codec slot. - -Thus, if the second error message appears, try to narrow the probed -codec slots via `probe_mask` option. It's a bitmask, and each bit -corresponds to the codec slot. For example, to probe only the first -slot, pass `probe_mask=1`. For the first and the third slots, pass -`probe_mask=5` (where 5 = 1 | 4), and so on. - -Since 2.6.29 kernel, the driver has a more robust probing method, so -this error might happen rarely, though. - -On a machine with a broken BIOS, sometimes you need to force the -driver to probe the codec slots the hardware doesn't report for use. -In such a case, turn the bit 8 (0x100) of `probe_mask` option on. -Then the rest 8 bits are passed as the codec slots to probe -unconditionally. For example, `probe_mask=0x103` will force to probe -the codec slots 0 and 1 no matter what the hardware reports. - - -Interrupt Handling -~~~~~~~~~~~~~~~~~~ -HD-audio driver uses MSI as default (if available) since 2.6.33 -kernel as MSI works better on some machines, and in general, it's -better for performance. However, Nvidia controllers showed bad -regressions with MSI (especially in a combination with AMD chipset), -thus we disabled MSI for them. - -There seem also still other devices that don't work with MSI. If you -see a regression wrt the sound quality (stuttering, etc) or a lock-up -in the recent kernel, try to pass `enable_msi=0` option to disable -MSI. If it works, you can add the known bad device to the blacklist -defined in hda_intel.c. In such a case, please report and give the -patch back to the upstream developer. - - -HD-AUDIO CODEC --------------- - -Model Option -~~~~~~~~~~~~ -The most common problem regarding the HD-audio driver is the -unsupported codec features or the mismatched device configuration. -Most of codec-specific code has several preset models, either to -override the BIOS setup or to provide more comprehensive features. - -The driver checks PCI SSID and looks through the static configuration -table until any matching entry is found. If you have a new machine, -you may see a message like below: ------------------------------------------------------------------------- - hda_codec: ALC880: BIOS auto-probing. ------------------------------------------------------------------------- -Meanwhile, in the earlier versions, you would see a message like: ------------------------------------------------------------------------- - hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... ------------------------------------------------------------------------- -Even if you see such a message, DON'T PANIC. Take a deep breath and -keep your towel. First of all, it's an informational message, no -warning, no error. This means that the PCI SSID of your device isn't -listed in the known preset model (white-)list. But, this doesn't mean -that the driver is broken. Many codec-drivers provide the automatic -configuration mechanism based on the BIOS setup. - -The HD-audio codec has usually "pin" widgets, and BIOS sets the default -configuration of each pin, which indicates the location, the -connection type, the jack color, etc. The HD-audio driver can guess -the right connection judging from these default configuration values. -However -- some codec-support codes, such as patch_analog.c, don't -support the automatic probing (yet as of 2.6.28). And, BIOS is often, -yes, pretty often broken. It sets up wrong values and screws up the -driver. - -The preset model 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. Passing 0 or a negative number to -the first or the second value will make the check of the corresponding -field be skipped. It'll be useful for really broken devices that don't -initialize SSID properly. - -The `[model]` line allows to change the model name of the each codec. -In the example above, it will be changed to model=auto. -Note that this overrides the module option. - -After the `[pincfg]` line, the contents are parsed as the initial -default pin-configurations just like `user_pin_configs` sysfs above. -The values can be shown in user_pin_configs sysfs file, too. - -Similarly, the lines after `[verb]` are parsed as `init_verbs` -sysfs entries, and the lines after `[hint]` are parsed as `hints` -sysfs entries, respectively. - -Another example to override the codec vendor id from 0x12345678 to -0xdeadbeef is like below: ------------------------------------------------------------------------- - [codec] - 0x12345678 0xabcd1234 2 - - [vendor_id] - 0xdeadbeef ------------------------------------------------------------------------- - -In the similar way, you can override the codec subsystem_id via -`[subsystem_id]`, the revision id via `[revision_id]` line. -Also, the codec chip name can be rewritten via `[chip_name]` line. ------------------------------------------------------------------------- - [codec] - 0x12345678 0xabcd1234 2 - - [subsystem_id] - 0xffff1111 - - [revision_id] - 0x10 - - [chip_name] - My-own NEWS-0002 ------------------------------------------------------------------------- - -The hd-audio driver reads the file via request_firmware(). Thus, -a patch file has to be located on the appropriate firmware path, -typically, /lib/firmware. For example, when you pass the option -`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be -present. - -The patch module option is specific to each card instance, and you -need to give one file name for each instance, separated by commas. -For example, if you have two cards, one for an on-board analog and one -for an HDMI video board, you may pass patch option like below: ------------------------------------------------------------------------- - options snd-hda-intel patch=on-board-patch,hdmi-patch ------------------------------------------------------------------------- - - -Power-Saving -~~~~~~~~~~~~ -The power-saving is a kind of auto-suspend of the device. When the -device is inactive for a certain time, the device is automatically -turned off to save the power. The time to go down is specified via -`power_save` module option, and this option can be changed dynamically -via sysfs. - -The power-saving won't work when the analog loopback is enabled on -some codecs. Make sure that you mute all unneeded signal routes when -you want the power-saving. - -The power-saving feature might cause audible click noises at each -power-down/up depending on the device. Some of them might be -solvable, but some are hard, I'm afraid. Some distros such as -openSUSE enables the power-saving feature automatically when the power -cable is unplugged. Thus, if you hear noises, suspect first the -power-saving. See /sys/module/snd_hda_intel/parameters/power_save to -check the current value. If it's non-zero, the feature is turned on. - - -Tracepoints -~~~~~~~~~~~ -The hd-audio driver gives a few basic tracepoints. -`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response` -traces the response from RIRB (only when read from the codec driver). -`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc, -`hda:hda_unsol_event` traces the unsolicited events, and -`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up -via power-saving behavior. - -Enabling all tracepoints can be done like ------------------------------------------------------------------------- - # echo 1 > /sys/kernel/debug/tracing/events/hda/enable ------------------------------------------------------------------------- -then after some commands, you can traces from -/sys/kernel/debug/tracing/trace file. For example, when you want to -trace what codec command is sent, enable the tracepoint like: ------------------------------------------------------------------------- - # cat /sys/kernel/debug/tracing/trace - # tracer: nop - # - # TASK-PID CPU# TIMESTAMP FUNCTION - # | | | | | - <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019 - <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019 - <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a - <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a - <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019 - <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019 - <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a - <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a ------------------------------------------------------------------------- -Here `[0:0]` indicates the card number and the codec address, and -`val` shows the value sent to the codec, respectively. The value is -a packed value, and you can decode it via hda-decode-verb program -included in hda-emu package below. For example, the value e3a019 is -to set the left output-amp value to 25. ------------------------------------------------------------------------- - % hda-decode-verb 0xe3a019 - raw value = 0x00e3a019 - cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19 - raw value: verb = 0x3a0, parm = 0x19 - verbname = set_amp_gain_mute - amp raw val = 0xa019 - output, left, idx=0, mute=0, val=25 ------------------------------------------------------------------------- - - -Development Tree -~~~~~~~~~~~~~~~~ -The latest development codes for HD-audio are found on sound git tree: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git - -The master branch or for-next branches can be used as the main -development branches in general while the 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.suse.com/pub/people/tiwai/snapshot/ - - -Sending a Bug Report -~~~~~~~~~~~~~~~~~~~~ -If any model or module options don't work for your device, it's time -to send a bug report to the developers. Give the following in your -bug report: - -- Hardware vendor, product and model names -- Kernel version (and ALSA-driver version if you built externally) -- `alsa-info.sh` output; run with `--no-upload` option. See the - section below about alsa-info - -If it's a regression, at best, send alsa-info outputs of both working -and non-working kernels. This is really helpful because we can -compare the codec registers directly. - -Send a bug report either the followings: - -kernel-bugzilla:: - https://bugzilla.kernel.org/ -alsa-devel ML:: - alsa-devel@alsa-project.org - - -DEBUG TOOLS ------------ - -This section describes some tools available for debugging HD-audio -problems. - -alsa-info -~~~~~~~~~ -The script `alsa-info.sh` is a very useful tool to gather the audio -device information. You can fetch the latest version from: - -- http://www.alsa-project.org/alsa-info.sh - -Run this script as root, and it will gather the important information -such as the module lists, module parameters, proc file contents -including the codec proc files, mixer outputs and the control -elements. As default, it will store the information onto a web server -on alsa-project.org. But, if you send a bug report, it'd be better to -run with `--no-upload` option, and attach the generated file. - -There are some other useful options. See `--help` option output for -details. - -When a probe error occurs or when the driver obviously assigns a -mismatched model, it'd be helpful to load the driver with -`probe_only=1` option (at best after the cold reboot) and run -alsa-info at this state. With this option, the driver won't configure -the mixer and PCM but just tries to probe the codec slot. After -probing, the proc file is available, so you can get the raw codec -information before modified by the driver. Of course, the driver -isn't usable with `probe_only=1`. But you can continue the -configuration via hwdep sysfs file if hda-reconfig option is enabled. -Using `probe_only` mask 2 skips the reset of HDA codecs (use -`probe_only=3` as module option). The hwdep interface can be used -to determine the BIOS codec initialization. - - -hda-verb -~~~~~~~~ -hda-verb is a tiny program that allows you to access the HD-audio -codec directly. You can execute a raw HD-audio codec verb with this. -This program accesses the hwdep device, thus you need to enable the -kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand. - -The hda-verb program takes four arguments: the hwdep device file, the -widget NID, the verb and the parameter. When you access to the codec -on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first -argument, typically. (However, the real path name depends on the -system.) - -The second parameter is the widget number-id to access. The third -parameter can be either a hex/digit number or a string corresponding -to a verb. Similarly, the last parameter is the value to write, or -can be a string for the parameter type. - ------------------------------------------------------------------------- - % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 - nid = 0x12, verb = 0x701, param = 0x2 - value = 0x0 - - % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID - nid = 0x0, verb = 0xf00, param = 0x0 - value = 0x10ec0262 - - % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 - nid = 0x2, verb = 0x300, param = 0xb080 - value = 0x0 ------------------------------------------------------------------------- - -Although you can issue any verbs with this program, the driver state -won't be always updated. For example, the volume values are usually -cached in the driver, and thus changing the widget amp value directly -via hda-verb won't change the mixer value. - -The hda-verb program is found in the ftp directory: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -Also a git repository is available: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git - -See README file in the tarball for more details about hda-verb -program. - - -hda-analyzer -~~~~~~~~~~~~ -hda-analyzer provides a graphical interface to access the raw HD-audio -control, based on pyGTK2 binding. It's a more powerful version of -hda-verb. The program gives you an easy-to-use GUI stuff for showing -the widget information and adjusting the amp values, as well as the -proc-compatible output. - -The hda-analyzer: - -- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer - -is a part of alsa.git repository in alsa-project.org: - -- git://git.alsa-project.org/alsa.git - -Codecgraph -~~~~~~~~~~ -Codecgraph is a utility program to generate a graph and visualizes the -codec-node connection of a codec chip. It's especially useful when -you analyze or debug a codec without a proper datasheet. The program -parses the given codec proc file and converts to SVG via graphiz -program. - -The tarball and GIT trees are found in the web page at: - -- http://helllabs.org/codecgraph/ - - -hda-emu -~~~~~~~ -hda-emu is an HD-audio emulator. The main purpose of this program is -to debug an HD-audio codec without the real hardware. Thus, it -doesn't emulate the behavior with the real audio I/O, but it just -dumps the codec register changes and the ALSA-driver internal changes -at probing and operating the HD-audio driver. - -The program requires a codec proc-file to simulate. Get a proc file -for the target codec beforehand, or pick up an example codec from the -codec proc collections in the tarball. Then, run the program with the -proc file, and the hda-emu program will start parsing the codec file -and simulates the HD-audio driver: - ------------------------------------------------------------------------- - % hda-emu codecs/stac9200-dell-d820-laptop - # Parsing.. - hda_codec: Unknown model for STAC9200, using BIOS defaults - hda_codec: pin nid 08 bios pin config 40c003fa - .... ------------------------------------------------------------------------- - -The program gives you only a very dumb command-line interface. You -can get a proc-file dump at the current state, get a list of control -(mixer) elements, set/get the control element value, simulate the PCM -operation, the jack plugging simulation, etc. - -The package is found in: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -A git repository is available: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git - -See README file in the tarball for more details about hda-emu -program. diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt deleted file mode 100644 index ccda41b10f8..00000000000 --- a/Documentation/sound/alsa/Joystick.txt +++ /dev/null @@ -1,86 +0,0 @@ -Analog Joystick Support on ALSA Drivers -======================================= - Oct. 14, 2003 - Takashi Iwai <tiwai@suse.de> - -General -------- - -First of all, you need to enable GAMEPORT support on Linux kernel for -using a joystick with the ALSA driver. For the details of gameport -support, refer to Documentation/input/joystick.txt. - -The joystick support of ALSA drivers is different between ISA and PCI -cards. In the case of ISA (PnP) cards, it's usually handled by the -independent module (ns558). Meanwhile, the ALSA PCI drivers have the -built-in gameport support. Hence, when the ALSA PCI driver is built -in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the -gameport support on that card will be (silently) disabled. - -Some adapter modules probe the physical connection of the device at -the load time. It'd be safer to plug in the joystick device before -loading the module. - - -PCI Cards ---------- - -For PCI cards, the joystick is enabled when the appropriate module -option is specified. Some drivers don't need options, and the -joystick support is always enabled. In the former ALSA version, there -was a dynamic control API for the joystick activation. It was -changed, however, to the static module options because of the system -stability and the resource management. - -The following PCI drivers support the joystick natively. - - Driver Module Option Available Values - --------------------------------------------------------------------------- - als4000 joystick_port 0 = disable (default), 1 = auto-detect, - manual: any address (e.g. 0x200) - au88x0 N/A N/A - azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default) - ens1370 joystick 0 = disable (default), 1 = enable - ens1371 joystick_port 0 = disable (default), 1 = auto-detect, - manual: 0x200, 0x208, 0x210, 0x218 - cmipci joystick_port 0 = disable (default), 1 = auto-detect, - manual: any address (e.g. 0x200) - cs4281 N/A N/A - cs46xx N/A N/A - es1938 N/A N/A - es1968 joystick 0 = disable (default), 1 = enable - sonicvibes N/A N/A - trident N/A N/A - via82xx(*1) joystick 0 = disable (default), 1 = enable - ymfpci joystick_port 0 = disable (default), 1 = auto-detect, - manual: 0x201, 0x202, 0x204, 0x205(*2) - --------------------------------------------------------------------------- - - *1) VIA686A/B only - *2) With YMF744/754 chips, the port address can be chosen arbitrarily - -The following drivers don't support gameport natively, but there are -additional modules. Load the corresponding module to add the gameport -support. - - Driver Additional Module - ----------------------------- - emu10k1 emu10k1-gp - fm801 fm801-gp - ----------------------------- - -Note: the "pcigame" and "cs461x" modules are for the OSS drivers only. - These ALSA drivers (cs46xx, trident and au88x0) have the - built-in gameport support. - -As mentioned above, ALSA PCI drivers have the built-in gameport -support, so you don't have to load ns558 module. Just load "joydev" -and the appropriate adapter module (e.g. "analog"). - - -ISA Cards ---------- - -ALSA ISA drivers don't have the built-in gameport support. -Instead, you need to load "ns558" module in addition to "joydev" and -the adapter module (e.g. "analog"). diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt deleted file mode 100644 index 4ee35b4fbe4..00000000000 --- a/Documentation/sound/alsa/MIXART.txt +++ /dev/null @@ -1,100 +0,0 @@ - Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards - Digigram <alsa@digigram.com> - - -GENERAL -======= - -The miXart8 is a multichannel audio processing and mixing soundcard -that has 4 stereo audio inputs and 4 stereo audio outputs. -The miXart8AES/EBU is the same with a add-on card that offers further -4 digital stereo audio inputs and outputs. -Furthermore the add-on card offers external clock synchronisation -(AES/EBU, Word Clock, Time Code and Video Synchro) - -The mainboard has a PowerPC that offers onboard mpeg encoding and -decoding, samplerate conversions and various effects. - -The driver don't work properly at all until the certain firmwares -are loaded, i.e. no PCM nor mixer devices will appear. -Use the mixartloader that can be found in the alsa-tools package. - - -VERSION 0.1.0 -============= - -One miXart8 board will be represented as 4 alsa cards, each with 1 -stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device. -With a miXart8AES/EBU there is in addition 1 stereo digital input -'pcm1c' and 1 stereo digital output 'pcm1p' per card. - -Formats -------- -U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE -Sample rates : 8000 - 48000 Hz continuously - -Playback --------- -For instance the playback devices are configured to have max. 4 -substreams performing hardware mixing. This could be changed to a -maximum of 24 substreams if wished. -Mono files will be played on the left and right channel. Each channel -can be muted for each stream to use 8 analog/digital outputs separately. - -Capture -------- -There is one substream per capture device. For instance only stereo -formats are supported. - -Mixer ------ -<Master> and <Master Capture> : analog volume control of playback and capture PCM. -<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream. -<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream. -<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume -and mute control. - -Rem : for best audio quality try to keep a 0 attenuation on the PCM -and AES volume controls which is set by 219 in the range from 0 to 255 -(about 86% with alsamixer) - - -NOT YET IMPLEMENTED -=================== - -- external clock support (AES/EBU, Word Clock, Time Code, Video Sync) -- MPEG audio formats -- mono record -- on-board effects and samplerate conversions -- linked streams - - -FIRMWARE -======== - -[As of 2.6.11, the firmware can be loaded automatically with hotplug - when CONFIG_FW_LOADER is set. The mixartloader is necessary only - for older versions or when you build the driver into kernel.] - -For loading the firmware automatically after the module is loaded, use a -install command. For example, add the following entry to -/etc/modprobe.d/mixart.conf for miXart driver: - - install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \ - /usr/bin/mixartloader -(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to - /etc/modules.conf, instead.) - -The firmware binaries are installed on /usr/share/alsa/firmware -(or /usr/local/share/alsa/firmware, depending to the prefix option of -configure). There will be a miXart.conf file, which define the dsp image -files. - -The firmware files are copyright by Digigram SA - - -COPYRIGHT -========= - -Copyright (c) 2003 Digigram SA <alsa@digigram.com> -Distributable under GPL. diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt deleted file mode 100644 index 152ca2a3f1b..00000000000 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ /dev/null @@ -1,305 +0,0 @@ - NOTES ON KERNEL OSS-EMULATION - ============================= - - Jan. 22, 2004 Takashi Iwai <tiwai@suse.de> - - -Modules -======= - -ALSA provides a powerful OSS emulation on the kernel. -The OSS emulation for PCM, mixer and sequencer devices is implemented -as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. -When you need to access the OSS PCM, mixer or sequencer devices, the -corresponding module has to be loaded. - -These modules are loaded automatically when the corresponding service -is called. The alias is defined sound-service-x-y, where x and y are -the card number and the minor unit number. Usually you don't have to -define these aliases by yourself. - -Only necessary step for auto-loading of OSS modules is to define the -card alias in /etc/modprobe.d/alsa.conf, such as - - alias sound-slot-0 snd-emu10k1 - -As the second card, define sound-slot-1 as well. -Note that you can't use the aliased name as the target name (i.e. -"alias sound-slot-0 snd-card-0" doesn't work any more like the old -modutils). - -The currently available OSS configuration is shown in -/proc/asound/oss/sndstat. This shows in the same syntax of -/dev/sndstat, which is available on the commercial OSS driver. -On ALSA, you can symlink /dev/sndstat to this proc file. - -Please note that the devices listed in this proc file appear only -after the corresponding OSS-emulation module is loaded. Don't worry -even if "NOT ENABLED IN CONFIG" is shown in it. - - -Device Mapping -============== - -ALSA supports the following OSS device files: - - PCM: - /dev/dspX - /dev/adspX - - Mixer: - /dev/mixerX - - MIDI: - /dev/midi0X - /dev/amidi0X - - Sequencer: - /dev/sequencer - /dev/sequencer2 (aka /dev/music) - -where X is the card number from 0 to 7. - -(NOTE: Some distributions have the device files like /dev/midi0 and - /dev/midi1. They are NOT for OSS but for tclmidi, which is - a totally different thing.) - -Unlike the real OSS, ALSA cannot use the device files more than the -assigned ones. For example, the first card cannot use /dev/dsp1 or -/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. - -As seen above, PCM and MIDI may have two devices. Usually, the first -PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary -device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and -/dev/amidi, respectively. - -You can change this device mapping via the module options of -snd-pcm-oss and snd-rawmidi. In the case of PCM, the following -options are available for snd-pcm-oss: - - dsp_map PCM device number assigned to /dev/dspX - (default = 0) - adsp_map PCM device number assigned to /dev/adspX - (default = 1) - -For example, to map the third PCM device (hw:0,2) to /dev/adsp0, -define like this: - - options snd-pcm-oss adsp_map=2 - -The options take arrays. For configuring the second card, specify -two entries separated by comma. For example, to map the third PCM -device on the second card to /dev/adsp1, define like below: - - options snd-pcm-oss adsp_map=0,2 - -To change the mapping of MIDI devices, the following options are -available for snd-rawmidi: - - midi_map MIDI device number assigned to /dev/midi0X - (default = 0) - amidi_map MIDI device number assigned to /dev/amidi0X - (default = 1) - -For example, to assign the third MIDI device on the first card to -/dev/midi00, define as follows: - - options snd-rawmidi midi_map=2 - - -PCM Mode -======== - -As default, ALSA emulates the OSS PCM with so-called plugin layer, -i.e. tries to convert the sample format, rate or channels -automatically when the card doesn't support it natively. -This will lead to some problems for some applications like quake or -wine, especially if they use the card only in the MMAP mode. - -In such a case, you can change the behavior of PCM per application by -writing a command to the proc file. There is a proc file for each PCM -stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number -(zero-based), Y the PCM device number (zero-based), and 'p' is for -playback and 'c' for capture, respectively. Note that this proc file -exists only after snd-pcm-oss module is loaded. - -The command sequence has the following syntax: - - app_name fragments fragment_size [options] - -app_name is the name of application with (higher priority) or without -path. -fragments specifies the number of fragments or zero if no specific -number is given. -fragment_size is the size of fragment in bytes or zero if not given. -options is the optional parameters. The following options are -available: - - disable the application tries to open a pcm device for - this channel but does not want to use it. - direct don't use plugins - block force block open mode - non-block force non-block open mode - partial-frag write also partial fragments (affects playback only) - no-silence do not fill silence ahead to avoid clicks - -The disable option is useful when one stream direction (playback or -capture) is not handled correctly by the application although the -hardware itself does support both directions. -The direct option is used, as mentioned above, to bypass the automatic -conversion and useful for MMAP-applications. -For example, to playback the first PCM device without plugins for -quake, send a command via echo like the following: - - % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss - -While quake wants only playback, you may append the second command -to notify driver that only this direction is about to be allocated: - - % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss - -The permission of proc files depend on the module options of snd. -As default it's set as root, so you'll likely need to be superuser for -sending the command above. - -The block and non-block options are used to change the behavior of -opening the device file. - -As default, ALSA behaves as original OSS drivers, i.e. does not block -the file when it's busy. The -EBUSY error is returned in this case. - -This blocking behavior can be changed globally via nonblock_open -module option of snd-pcm-oss. For using the blocking mode as default -for OSS devices, define like the following: - - options snd-pcm-oss nonblock_open=0 - -The partial-frag and no-silence commands have been added recently. -Both commands are for optimization use only. The former command -specifies to invoke the write transfer only when the whole fragment is -filled. The latter stops writing the silence data ahead -automatically. Both are disabled as default. - -You can check the currently defined configuration by reading the proc -file. The read image can be sent to the proc file again, hence you -can save the current configuration - - % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg - -and restore it like - - % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss - -Also, for clearing all the current configuration, send "erase" command -as below: - - % echo "erase" > /proc/asound/card0/pcm0p/oss - - -Mixer Elements -============== - -Since ALSA has completely different mixer interface, the emulation of -OSS mixer is relatively complicated. ALSA builds up a mixer element -from several different ALSA (mixer) controls based on the name -string. For example, the volume element SOUND_MIXER_PCM is composed -from "PCM Playback Volume" and "PCM Playback Switch" controls for the -playback direction and from "PCM Capture Volume" and "PCM Capture -Switch" for the capture directory (if exists). When the PCM volume of -OSS is changed, all the volume and switch controls above are adjusted -automatically. - -As default, ALSA uses the following control for OSS volumes: - - OSS volume ALSA control Index - ----------------------------------------------------- - SOUND_MIXER_VOLUME Master 0 - SOUND_MIXER_BASS Tone Control - Bass 0 - SOUND_MIXER_TREBLE Tone Control - Treble 0 - SOUND_MIXER_SYNTH Synth 0 - SOUND_MIXER_PCM PCM 0 - SOUND_MIXER_SPEAKER PC Speaker 0 - SOUND_MIXER_LINE Line 0 - SOUND_MIXER_MIC Mic 0 - SOUND_MIXER_CD CD 0 - SOUND_MIXER_IMIX Monitor Mix 0 - SOUND_MIXER_ALTPCM PCM 1 - SOUND_MIXER_RECLEV (not assigned) - SOUND_MIXER_IGAIN Capture 0 - SOUND_MIXER_OGAIN Playback 0 - SOUND_MIXER_LINE1 Aux 0 - SOUND_MIXER_LINE2 Aux 1 - SOUND_MIXER_LINE3 Aux 2 - SOUND_MIXER_DIGITAL1 Digital 0 - SOUND_MIXER_DIGITAL2 Digital 1 - SOUND_MIXER_DIGITAL3 Digital 2 - SOUND_MIXER_PHONEIN Phone 0 - SOUND_MIXER_PHONEOUT Phone 1 - SOUND_MIXER_VIDEO Video 0 - SOUND_MIXER_RADIO Radio 0 - SOUND_MIXER_MONITOR Monitor 0 - -The second column is the base-string of the corresponding ALSA -control. In fact, the controls with "XXX [Playback|Capture] -[Volume|Switch]" will be checked in addition. - -The current assignment of these mixer elements is listed in the proc -file, /proc/asound/cardX/oss_mixer, which will be like the following - - VOLUME "Master" 0 - BASS "" 0 - TREBLE "" 0 - SYNTH "" 0 - PCM "PCM" 0 - ... - -where the first column is the OSS volume element, the second column -the base-string of the corresponding ALSA control, and the third the -control index. When the string is empty, it means that the -corresponding OSS control is not available. - -For changing the assignment, you can write the configuration to this -proc file. For example, to map "Wave Playback" to the PCM volume, -send the command like the following: - - % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer - -The command is exactly as same as listed in the proc file. You can -change one or more elements, one volume per line. In the last -example, both "Wave Playback Volume" and "Wave Playback Switch" will -be affected when PCM volume is changed. - -Like the case of PCM proc file, the permission of proc files depend on -the module options of snd. you'll likely need to be superuser for -sending the command above. - -As well as in the case of PCM proc file, you can save and restore the -current mixer configuration by reading and writing the whole file -image. - - -Duplex Streams -============== - -Note that when attempting to use a single device file for playback and -capture, the OSS API provides no way to set the format, sample rate or -number of channels different in each direction. Thus - io_handle = open("device", O_RDWR) -will only function correctly if the values are the same in each direction. - -To use different values in the two directions, use both - input_handle = open("device", O_RDONLY) - output_handle = open("device", O_WRONLY) -and set the values for the corresponding handle. - - -Unsupported Features -==================== - -MMAP on ICE1712 driver ----------------------- -ICE1712 supports only the unconventional format, interleaved -10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap -the buffer as the conventional (mono or 2-channels, 8 or 16bit) format -on OSS. - diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt deleted file mode 100644 index 7fcd1ad96fc..00000000000 --- a/Documentation/sound/alsa/Procfile.txt +++ /dev/null @@ -1,243 +0,0 @@ - Proc Files of ALSA Drivers - ========================== - Takashi Iwai <tiwai@suse.de> - -General -------- - -ALSA has its own proc tree, /proc/asound. Many useful information are -found in this tree. When you encounter a problem and need debugging, -check the files listed in the following sections. - -Each card has its subtree cardX, where X is from 0 to 7. The -card-specific files are stored in the card* subdirectories. - - -Global Information ------------------- - -cards - Shows the list of currently configured ALSA drivers, - index, the id string, short and long descriptions. - -version - Shows the version string and compile date. - -modules - Lists the module of each card - -devices - Lists the ALSA native device mappings. - -meminfo - Shows the status of allocated pages via ALSA drivers. - Appears only when CONFIG_SND_DEBUG=y. - -hwdep - Lists the currently available hwdep devices in format of - <card>-<device>: <name> - -pcm - Lists the currently available PCM devices in format of - <card>-<device>: <id>: <name> : <sub-streams> - -timer - Lists the currently available timer devices - - -oss/devices - Lists the OSS device mappings. - -oss/sndstat - Provides the output compatible with /dev/sndstat. - You can symlink this to /dev/sndstat. - - -Card Specific Files -------------------- - -The card-specific files are found in /proc/asound/card* directories. -Some drivers (e.g. cmipci) have their own proc entries for the -register dump, etc (e.g. /proc/asound/card*/cmipci shows the register -dump). These files would be really helpful for debugging. - -When PCM devices are available on this card, you can see directories -like pcm0p or pcm1c. They hold the PCM information for each PCM -stream. The number after 'pcm' is the PCM device number from 0, and -the last 'p' or 'c' means playback or capture direction. The files in -this subtree is described later. - -The status of MIDI I/O is found in midi* files. It shows the device -name and the received/transmitted bytes through the MIDI device. - -When the card is equipped with AC97 codecs, there are codec97#* -subdirectories (described later). - -When the OSS mixer emulation is enabled (and the module is loaded), -oss_mixer file appears here, too. This shows the current mapping of -OSS mixer elements to the ALSA control elements. You can change the -mapping by writing to this device. Read OSS-Emulation.txt for -details. - - -PCM Proc Files --------------- - -card*/pcm*/info - The general information of this PCM device: card #, device #, - substreams, etc. - -card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y and - CONFIG_PCM_XRUN_DEBUG=y. - This shows the status of xrun (= buffer overrun/xrun) and - invalid PCM position debug/check of ALSA PCM middle layer. - It takes an integer value, can be changed by writing to this - file, such as - - # echo 5 > /proc/asound/card0/pcm0p/xrun_debug - - The value consists of the following bit flags: - bit 0 = Enable XRUN/jiffies debug messages - bit 1 = Show stack trace at XRUN / jiffies check - bit 2 = Enable additional jiffies check - 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 deleted file mode 100644 index 0e41576fa13..00000000000 --- a/Documentation/sound/alsa/README.maya44 +++ /dev/null @@ -1,163 +0,0 @@ -NOTE: The following is the original document of Rainer's patch that the -current maya44 code based on. Some contents might be obsoleted, but I -keep here as reference -- tiwai - ----------------------------------------------------------------- - -STATE OF DEVELOPMENT: - -This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. -Development is carried out by Rainer Zimmermann (mail@lightshed.de). - -ESI provided a sample Maya44 card for the development work. - -However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing. - -This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008). - - -The following functions work, as tested by Rainer Zimmermann and Piotr Makowski: - -- playback and capture at all sampling rates -- input/output level -- crossmixing -- line/mic switch -- phantom power switch -- analogue monitor a.k.a bypass - - -The following functions *should* work, but are not fully tested: - -- Channel 3+4 analogue - S/PDIF input switching -- S/PDIF output -- all inputs/outputs on the M/IO/DIO extension card -- internal/external clock selection - - -*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.* - - -Things that do not seem to work: - -- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code). - -- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down. - - -DRIVER DETAILS: - -the following files were added: - -pci/ice1724/maya44.c - Maya44 specific code -pci/ice1724/maya44.h -pci/ice1724/ice1724.patch -pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) -i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs -include/wm8776.h - - -Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. -This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately. - - -the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree: - -wtm.h -vt1720_mobo.h -revo.h -prodigy192.h -pontis.h -phase.h -maya44.h -juli.h -aureon.h -amp.h -envy24ht.h -se.h -prodigy_hifi.h - - -*I hope this is the correct way to do things.* - - -SAMPLING RATES: - -The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture. - -As the ICE1724 chip only allows one global sampling rate, this is handled as follows: - -* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels. - -* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices. - -*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality. - - -I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic. - -The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). - - -SOUND DEVICES: - -PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0): - -hw:0,0 input - stereo, analog input 1+2 -hw:0,0 output - stereo, analog output 1+2 -hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input -hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) - - -NAMING OF MIXER CONTROLS: - -(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software). - - -PCM: (digital) output level for channel 1+2 -PCM 1: same for channel 3+4 - -Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. - Make sure this is not turned on while any other source is connected to input 1/2. - It might damage the source and/or the maya44 card. - -Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). - -Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. -Bypass 1: same for channel 3+4. - -Crossmix: cross-mixer from channels 1+2 to channels 3+4 -Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 - -IEC958 Output: switch for S/PDIF output. - This is not supported by the ESI windows driver. - S/PDIF should output the same signal as channel 3+4. [untested!] - - -Digitial output selectors: - - These switches allow a direct digital routing from the ADCs to the DACs. - Each switch determines where the digital input data to one of the DACs comes from. - They are not supported by the ESI windows driver. - For normal operation, they should all be set to "PCM out". - -H/W: Output source channel 1 -H/W 1: Output source channel 2 -H/W 2: Output source channel 3 -H/W 3: Output source channel 4 - -H/W 4 ... H/W 9: unknown function, left in to enable testing. - Possibly some of these control S/PDIF output(s). - If these turn out to be unused, they will go away in later driver versions. - -Selectable values for each of the digital output selectors are: - "PCM out" -> DAC output of the corresponding channel (default setting) - "Input 1"... - "Input 4" -> direct routing from ADC output of the selected input channel - - --------- - -Feb 14, 2008 -Rainer Zimmermann -mail@lightshed.de - diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt deleted file mode 100644 index f4b5988f450..00000000000 --- a/Documentation/sound/alsa/SB-Live-mixer.txt +++ /dev/null @@ -1,356 +0,0 @@ - - Sound Blaster Live mixer / default DSP code - =========================================== - - -The EMU10K1 chips have a DSP part which can be programmed to support -various ways of sample processing, which is described here. -(This article does not deal with the overall functionality of the -EMU10K1 chips. See the manuals section for further details.) - -The ALSA driver programs this portion of chip by default code -(can be altered later) which offers the following functionality: - - -1) IEC958 (S/PDIF) raw PCM --------------------------- - -This PCM device (it's the 4th PCM device (index 3!) and first subdevice -(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit -little endian streams without any modifications to the digital output -(coaxial or optical). The universal interface allows the creation of up -to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would -be easy to add support for multichannel devices to the current code, -but the conversion routines exist only for stereo (2-channel streams) -at the time. - -Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details. - - -2) Digital mixer controls -------------------------- - -These controls are built using the DSP instructions. They offer extended -functionality. Only the default build-in code in the ALSA driver is described -here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). - - -Explanation of used abbreviations: - -DAC - digital to analog converter -ADC - analog to digital converter -I2S - one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) -LFE - low frequency effects (subwoofer signal) -AC97 - a chip containing an analog mixer, DAC and ADC converters -IEC958 - S/PDIF -FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators. - Each of the synthesizer voices can feed its output to these accumulators - and the DSP microcontroller can operate with the resulting sum. - - -name='Wave Playback Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. -The result samples are forwarded to the front DAC PCM slots of the AC97 codec. - -name='Wave Surround Playback Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. -The result samples are forwarded to the rear I2S DACs. These DACs operates -separately (they are not inside the AC97 codec). - -name='Wave Center Playback Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples. -The result is mixed to mono signal (single channel) and forwarded to -the ??rear?? right DAC PCM slot of the AC97 codec. - -name='Wave LFE Playback Volume',index=0 - -This control is used to attenuate samples for left and right PCM FX-bus -accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. -The result is mixed to mono signal (single channel) and forwarded to -the ??rear?? left DAC PCM slot of the AC97 codec. - -name='Wave Capture Volume',index=0 -name='Wave Capture Switch',index=0 - -These controls are used to attenuate samples for left and right PCM FX-bus -accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Synth Playback Volume',index=0 - -This control is used to attenuate samples for left and right MIDI FX-bus -accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. -The result samples are forwarded to the front DAC PCM slots of the AC97 codec. - -name='Synth Capture Volume',index=0 -name='Synth Capture Switch',index=0 - -These controls are used to attenuate samples for left and right MIDI FX-bus -accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Surround Playback Volume',index=0 - -This control is used to attenuate samples for left and right rear PCM FX-bus -accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. -The result samples are forwarded to the rear I2S DACs. These DACs operate -separately (they are not inside the AC97 codec). - -name='Surround Capture Volume',index=0 -name='Surround Capture Switch',index=0 - -These controls are used to attenuate samples for left and right rear PCM FX-bus -accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples. -The result is forwarded to the ADC capture FIFO (thus to the standard capture -PCM device). - -name='Center Playback Volume',index=0 - -This control is used to attenuate sample for center PCM FX-bus accumulator. -ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded -to the ??rear?? right DAC PCM slot of the AC97 codec. - -name='LFE Playback Volume',index=0 - -This control is used to attenuate sample for center PCM FX-bus accumulator. -ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded -to the ??rear?? left DAC PCM slot of the AC97 codec. - -name='AC97 Playback Volume',index=0 - -This control is used to attenuate samples for left and right front ADC PCM slots -of the AC97 codec. The result samples are forwarded to the front DAC PCM -slots of the AC97 codec. -******************************************************************************** -*** Note: This control should be zero for the standard operations, otherwise *** -*** a digital loopback is activated. *** -******************************************************************************** - -name='AC97 Capture Volume',index=0 - -This control is used to attenuate samples for left and right front ADC PCM slots -of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to -the standard capture PCM device). -******************************************************************************** -*** Note: This control should be 100 (maximal value), otherwise no analog *** -*** inputs of the AC97 codec can be captured (recorded). *** -******************************************************************************** - -name='IEC958 TTL Playback Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 TTL -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the front DAC PCM slots of the AC97 codec. - -name='IEC958 TTL Capture Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 TTL -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the ADC capture FIFO (thus to the standard capture PCM device). - -name='Zoom Video Playback Volume',index=0 - -This control is used to attenuate samples from left and right zoom video -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the front DAC PCM slots of the AC97 codec. - -name='Zoom Video Capture Volume',index=0 - -This control is used to attenuate samples from left and right zoom video -digital inputs (usually used by a CDROM drive). The result samples are -forwarded to the ADC capture FIFO (thus to the standard capture PCM device). - -name='IEC958 LiveDrive Playback Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 optical -digital input. The result samples are forwarded to the front DAC PCM slots -of the AC97 codec. - -name='IEC958 LiveDrive Capture Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 optical -digital inputs. The result samples are forwarded to the ADC capture FIFO -(thus to the standard capture PCM device). - -name='IEC958 Coaxial Playback Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 coaxial -digital inputs. The result samples are forwarded to the front DAC PCM slots -of the AC97 codec. - -name='IEC958 Coaxial Capture Volume',index=0 - -This control is used to attenuate samples from left and right IEC958 coaxial -digital inputs. The result samples are forwarded to the ADC capture FIFO -(thus to the standard capture PCM device). - -name='Line LiveDrive Playback Volume',index=0 -name='Line LiveDrive Playback Volume',index=1 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the LiveDrive). The result samples are forwarded to the front -DAC PCM slots of the AC97 codec. - -name='Line LiveDrive Capture Volume',index=1 -name='Line LiveDrive Capture Volume',index=1 - -This control is used to attenuate samples from left and right I2S ADC -inputs (on the LiveDrive). The result samples are forwarded to the ADC -capture FIFO (thus to the standard capture PCM device). - -name='Tone Control - Switch',index=0 - -This control turns the tone control on or off. The samples for front, rear -and center / LFE outputs are affected. - -name='Tone Control - Bass',index=0 - -This control sets the bass intensity. There is no neutral value!! -When the tone control code is activated, the samples are always modified. -The closest value to pure signal is 20. - -name='Tone Control - Treble',index=0 - -This control sets the treble intensity. There is no neutral value!! -When the tone control code is activated, the samples are always modified. -The closest value to pure signal is 20. - -name='IEC958 Optical Raw Playback Switch',index=0 - -If this switch is on, then the samples for the IEC958 (S/PDIF) digital -output are taken only from the raw FX8010 PCM, otherwise standard front -PCM samples are taken. - -name='Headphone Playback Volume',index=1 - -This control attenuates the samples for the headphone output. - -name='Headphone Center Playback Switch',index=1 - -If this switch is on, then the sample for the center PCM is put to the -left headphone output (useful for SB Live cards without separate center/LFE -output). - -name='Headphone LFE Playback Switch',index=1 - -If this switch is on, then the sample for the center PCM is put to the -right headphone output (useful for SB Live cards without separate center/LFE -output). - - -3) PCM stream related controls ------------------------------- - -name='EMU10K1 PCM Volume',index 0-31 - -Channel volume attenuation in range 0-0xffff. The maximum value (no -attenuation) is default. The channel mapping for three values is -as follows: - - 0 - mono, default 0xffff (no attenuation) - 1 - left, default 0xffff (no attenuation) - 2 - right, default 0xffff (no attenuation) - -name='EMU10K1 PCM Send Routing',index 0-31 - -This control specifies the destination - FX-bus accumulators. There are -twelve values with this mapping: - - 0 - mono, A destination (FX-bus 0-15), default 0 - 1 - mono, B destination (FX-bus 0-15), default 1 - 2 - mono, C destination (FX-bus 0-15), default 2 - 3 - mono, D destination (FX-bus 0-15), default 3 - 4 - left, A destination (FX-bus 0-15), default 0 - 5 - left, B destination (FX-bus 0-15), default 1 - 6 - left, C destination (FX-bus 0-15), default 2 - 7 - left, D destination (FX-bus 0-15), default 3 - 8 - right, A destination (FX-bus 0-15), default 0 - 9 - right, B destination (FX-bus 0-15), default 1 - 10 - right, C destination (FX-bus 0-15), default 2 - 11 - right, D destination (FX-bus 0-15), default 3 - -Don't forget that it's illegal to assign a channel to the same FX-bus accumulator -more than once (it means 0=0 && 1=0 is an invalid combination). - -name='EMU10K1 PCM Send Volume',index 0-31 - -It specifies the attenuation (amount) for given destination in range 0-255. -The channel mapping is following: - - 0 - mono, A destination attn, default 255 (no attenuation) - 1 - mono, B destination attn, default 255 (no attenuation) - 2 - mono, C destination attn, default 0 (mute) - 3 - mono, D destination attn, default 0 (mute) - 4 - left, A destination attn, default 255 (no attenuation) - 5 - left, B destination attn, default 0 (mute) - 6 - left, C destination attn, default 0 (mute) - 7 - left, D destination attn, default 0 (mute) - 8 - right, A destination attn, default 0 (mute) - 9 - right, B destination attn, default 255 (no attenuation) - 10 - right, C destination attn, default 0 (mute) - 11 - right, D destination attn, default 0 (mute) - - - -4) MANUALS/PATENTS: -------------------- - -ftp://opensource.creative.com/pub/doc -------------------------------------- - - Files: - LM4545.pdf AC97 Codec - - m2049.pdf The EMU10K1 Digital Audio Processor - - hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects - - -WIPO Patents ------------- - Patent numbers: - WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999) - streams - - WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999) - - WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction - Execution and Audio Data Sequencing (Jan. 14, 1999) - - -US Patents (http://www.uspto.gov/) ----------------------------------- - - US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) - - US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999) - with a multiport memory onto which multiple asynchronous - digital sound samples can be concurrently loaded - - US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999) - - US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000) - - US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000) - system bus with prioritization and modification of bus transfers - in accordance with loop ends and minimum block sizes - - US 6151670 Method for conserving memory storage using a (Nov. 21, 2000) - pool of short term memory registers - - US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001) - a common interrupt by associating programs to GP registers, - defining interrupt register, polling GP registers, and invoking - callback routine associated with defined interrupt register diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt deleted file mode 100644 index 1b0ac06ba95..00000000000 --- a/Documentation/sound/alsa/VIA82xx-mixer.txt +++ /dev/null @@ -1,8 +0,0 @@ - - VIA82xx mixer - ============= - -On many VIA82xx boards, the 'Input Source Select' mixer control does not work. -Setting it to 'Input2' on such boards will cause recording to hang, or fail -with EIO (input/output error) via OSS emulation. This control should be left -at 'Input1' for such cards. diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt deleted file mode 100644 index 0fa40679b08..00000000000 --- a/Documentation/sound/alsa/alsa-parameters.txt +++ /dev/null @@ -1,135 +0,0 @@ - ALSA Kernel Parameters - ~~~~~~~~~~~~~~~~~~~~~~ - -See Documentation/kernel-parameters.txt for general information on -specifying module parameters. - -This document may not be entirely up to date and comprehensive. The command -"modinfo -p ${modulename}" shows a current list of all parameters of a loadable -module. Loadable modules, after being loaded into the running kernel, also -reveal their parameters in /sys/module/${modulename}/parameters/. Some of these -parameters may be changed at runtime by the command -"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". - - - snd-ad1816a= [HW,ALSA] - - snd-ad1848= [HW,ALSA] - - snd-ali5451= [HW,ALSA] - - snd-als100= [HW,ALSA] - - snd-als4000= [HW,ALSA] - - snd-azt2320= [HW,ALSA] - - snd-cmi8330= [HW,ALSA] - - snd-cmipci= [HW,ALSA] - - snd-cs4231= [HW,ALSA] - - snd-cs4232= [HW,ALSA] - - snd-cs4236= [HW,ALSA] - - snd-cs4281= [HW,ALSA] - - snd-cs46xx= [HW,ALSA] - - snd-dt019x= [HW,ALSA] - - snd-dummy= [HW,ALSA] - - snd-emu10k1= [HW,ALSA] - - snd-ens1370= [HW,ALSA] - - snd-ens1371= [HW,ALSA] - - snd-es968= [HW,ALSA] - - snd-es1688= [HW,ALSA] - - snd-es18xx= [HW,ALSA] - - snd-es1938= [HW,ALSA] - - snd-es1968= [HW,ALSA] - - snd-fm801= [HW,ALSA] - - snd-gusclassic= [HW,ALSA] - - snd-gusextreme= [HW,ALSA] - - snd-gusmax= [HW,ALSA] - - snd-hdsp= [HW,ALSA] - - snd-ice1712= [HW,ALSA] - - snd-intel8x0= [HW,ALSA] - - snd-interwave= [HW,ALSA] - - snd-interwave-stb= - [HW,ALSA] - - snd-korg1212= [HW,ALSA] - - snd-maestro3= [HW,ALSA] - - snd-mpu401= [HW,ALSA] - - snd-mtpav= [HW,ALSA] - - snd-nm256= [HW,ALSA] - - snd-opl3sa2= [HW,ALSA] - - snd-opti92x-ad1848= - [HW,ALSA] - - snd-opti92x-cs4231= - [HW,ALSA] - - snd-opti93x= [HW,ALSA] - - snd-pmac= [HW,ALSA] - - snd-rme32= [HW,ALSA] - - snd-rme96= [HW,ALSA] - - snd-rme9652= [HW,ALSA] - - snd-sb8= [HW,ALSA] - - snd-sb16= [HW,ALSA] - - snd-sbawe= [HW,ALSA] - - snd-serial= [HW,ALSA] - - snd-sgalaxy= [HW,ALSA] - - snd-sonicvibes= [HW,ALSA] - - snd-sun-amd7930= - [HW,ALSA] - - snd-sun-cs4231= [HW,ALSA] - - snd-trident= [HW,ALSA] - - snd-usb-audio= [HW,ALSA,USB] - - snd-via82xx= [HW,ALSA] - - snd-virmidi= [HW,ALSA] - - snd-wavefront= [HW,ALSA] - - snd-ymfpci= [HW,ALSA] diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt deleted file mode 100644 index 90e9b3a11ab..00000000000 --- a/Documentation/sound/alsa/compress_offload.txt +++ /dev/null @@ -1,188 +0,0 @@ - compress_offload.txt - ===================== - Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com> - Vinod Koul <vinod.koul@linux.intel.com> - -Overview - -Since its early days, the ALSA API was defined with PCM support or -constant bitrates payloads such as IEC61937 in mind. Arguments and -returned values in frames are the norm, making it a challenge to -extend the existing API to compressed data streams. - -In recent years, audio digital signal processors (DSP) were integrated -in system-on-chip designs, and DSPs are also integrated in audio -codecs. Processing compressed data on such DSPs results in a dramatic -reduction of power consumption compared to host-based -processing. Support for such hardware has not been very good in Linux, -mostly because of a lack of a generic API available in the mainline -kernel. - -Rather than requiring a compatibility break with an API change of the -ALSA PCM interface, a new 'Compressed Data' API is introduced to -provide a control and data-streaming interface for audio DSPs. - -The design of this API was inspired by the 2-year experience with the -Intel Moorestown SOC, with many corrections required to upstream the -API in the mainline kernel instead of the staging tree and make it -usable by others. - -Requirements - -The main requirements are: - -- separation between byte counts and time. Compressed formats may have - a header per file, per frame, or no header at all. The payload size - may vary from frame-to-frame. As a result, it is not possible to - estimate reliably the duration of audio buffers when handling - compressed data. Dedicated mechanisms are required to allow for - reliable audio-video synchronization, which requires precise - reporting of the number of samples rendered at any given time. - -- Handling of multiple formats. PCM data only requires a specification - of the sampling rate, number of channels and bits per sample. In - contrast, compressed data comes in a variety of formats. Audio DSPs - may also provide support for a limited number of audio encoders and - decoders embedded in firmware, or may support more choices through - dynamic download of libraries. - -- Focus on main formats. This API provides support for the most - popular formats used for audio and video capture and playback. It is - likely that as audio compression technology advances, new formats - will be added. - -- Handling of multiple configurations. Even for a given format like - AAC, some implementations may support AAC multichannel but HE-AAC - stereo. Likewise WMA10 level M3 may require too much memory and cpu - cycles. The new API needs to provide a generic way of listing these - formats. - -- Rendering/Grabbing only. This API does not provide any means of - hardware acceleration, where PCM samples are provided back to - user-space for additional processing. This API focuses instead on - streaming compressed data to a DSP, with the assumption that the - decoded samples are routed to a physical output or logical back-end. - - - Complexity hiding. Existing user-space multimedia frameworks all - have existing enums/structures for each compressed format. This new - API assumes the existence of a platform-specific compatibility layer - to expose, translate and make use of the capabilities of the audio - DSP, eg. Android HAL or PulseAudio sinks. By construction, regular - applications are not supposed to make use of this API. - - -Design - -The new API shares a number of concepts with with the PCM API for flow -control. Start, pause, resume, drain and stop commands have the same -semantics no matter what the content is. - -The concept of memory ring buffer divided in a set of fragments is -borrowed from the ALSA PCM API. However, only sizes in bytes can be -specified. - -Seeks/trick modes are assumed to be handled by the host. - -The notion of rewinds/forwards is not supported. Data committed to the -ring buffer cannot be invalidated, except when dropping all buffers. - -The Compressed Data API does not make any assumptions on how the data -is transmitted to the audio DSP. DMA transfers from main memory to an -embedded audio cluster or to a SPI interface for external DSPs are -possible. As in the ALSA PCM case, a core set of routines is exposed; -each driver implementer will have to write support for a set of -mandatory routines and possibly make use of optional ones. - -The main additions are - -- get_caps -This routine returns the list of audio formats supported. Querying the -codecs on a capture stream will return encoders, decoders will be -listed for playback streams. - -- get_codec_caps For each codec, this routine returns a list of -capabilities. The intent is to make sure all the capabilities -correspond to valid settings, and to minimize the risks of -configuration failures. For example, for a complex codec such as AAC, -the number of channels supported may depend on a specific profile. If -the capabilities were exposed with a single descriptor, it may happen -that a specific combination of profiles/channels/formats may not be -supported. Likewise, embedded DSPs have limited memory and cpu cycles, -it is likely that some implementations make the list of capabilities -dynamic and dependent on existing workloads. In addition to codec -settings, this routine returns the minimum buffer size handled by the -implementation. This information can be a function of the DMA buffer -sizes, the number of bytes required to synchronize, etc, and can be -used by userspace to define how much needs to be written in the ring -buffer before playback can start. - -- set_params -This routine sets the configuration chosen for a specific codec. The -most important field in the parameters is the codec type; in most -cases decoders will ignore other fields, while encoders will strictly -comply to the settings - -- get_params -This routines returns the actual settings used by the DSP. Changes to -the settings should remain the exception. - -- get_timestamp -The timestamp becomes a multiple field structure. It lists the number -of bytes transferred, the number of samples processed and the number -of samples rendered/grabbed. All these values can be used to determine -the avarage bitrate, figure out if the ring buffer needs to be -refilled or the delay due to decoding/encoding/io on the DSP. - -Note that the list of codecs/profiles/modes was derived from the -OpenMAX AL specification instead of reinventing the wheel. -Modifications include: -- Addition of FLAC and IEC formats -- Merge of encoder/decoder capabilities -- Profiles/modes listed as bitmasks to make descriptors more compact -- Addition of set_params for decoders (missing in OpenMAX AL) -- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL) -- Addition of format information for WMA -- Addition of encoding options when required (derived from OpenMAX IL) -- Addition of rateControlSupported (missing in OpenMAX AL) - -Not supported: - -- Support for VoIP/circuit-switched calls is not the target of this - API. Support for dynamic bit-rate changes would require a tight - coupling between the DSP and the host stack, limiting power savings. - -- Packet-loss concealment is not supported. This would require an - additional interface to let the decoder synthesize data when frames - are lost during transmission. This may be added in the future. - -- Volume control/routing is not handled by this API. Devices exposing a - compressed data interface will be considered as regular ALSA devices; - volume changes and routing information will be provided with regular - ALSA kcontrols. - -- Embedded audio effects. Such effects should be enabled in the same - manner, no matter if the input was PCM or compressed. - -- multichannel IEC encoding. Unclear if this is required. - -- Encoding/decoding acceleration is not supported as mentioned - above. It is possible to route the output of a decoder to a capture - stream, or even implement transcoding capabilities. This routing - would be enabled with ALSA kcontrols. - -- Audio policy/resource management. This API does not provide any - hooks to query the utilization of the audio DSP, nor any premption - mechanisms. - -- No notion of underun/overrun. Since the bytes written are compressed - in nature and data written/read doesn't translate directly to - rendered output in time, this does not deal with underrun/overun and - maybe dealt in user-library - -Credits: -- Mark Brown and Liam Girdwood for discussions on the need for this API -- Harsha Priya for her work on intel_sst compressed API -- Rakesh Ughreja for valuable feedback -- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for - demonstrating and quantifying the benefits of audio offload on a - real platform. diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt deleted file mode 100644 index 751d45036a0..00000000000 --- a/Documentation/sound/alsa/emu10k1-jack.txt +++ /dev/null @@ -1,74 +0,0 @@ -This document is a guide to using the emu10k1 based devices with JACK for low -latency, multichannel recording functionality. All of my recent work to allow -Linux users to use the full capabilities of their hardware has been inspired -by the kX Project. Without their work I never would have discovered the true -power of this hardware. - - http://www.kxproject.com - - Lee Revell, 2005.03.30 - -Low latency, multichannel audio with JACK and the emu10k1/emu10k2 ------------------------------------------------------------------ - -Until recently, emu10k1 users on Linux did not have access to the same low -latency, multichannel features offered by the "kX ASIO" feature of their -Windows driver. As of ALSA 1.0.9 this is no more! - -For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback -channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or -even 32 (0.66ms) frames should work well. - -The configuration is slightly more involved than on Windows, as you have to -select the correct device for JACK to use. Actually, for qjackctl users it's -fairly self explanatory - select Duplex, then for capture and playback select -the multichannel devices, set the in and out channels to 16, and the sample -rate to 48000Hz. The command line looks like this: - -/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S - -This will give you 16 input ports and 16 output ports. - -The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the -Audigy). The mapping from FX bus to physical output is described in -SB-Live-mixer.txt (or Audigy-mixer.txt). - -The 16 input ports are connected to the 16 physical inputs. Contrary to -popular belief, all emu10k1 cards are multichannel cards. Which of these -input channels have physical inputs connected to them depends on the card -model. Trial and error is highly recommended; the pinout diagrams -for the card have been reverse engineered by some enterprising kX users and are -available on the internet. Meterbridge is helpful here, and the kX forums are -packed with useful information. - -Each input port will either correspond to a digital (SPDIF) input, an analog -input, or nothing. The one exception is the SBLive! 5.1. On these devices, -the second and third input ports are wired to the center/LFE output. You will -still see 16 capture channels, but only 14 are available for recording inputs. - -This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK -ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output) -channels. - -/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards: --------------------------------------------- -JACK Epilog FXBUS2(nr) --------------------------------------------- -capture_1 asio14 FXBUS2(0xe) -capture_2 asio15 FXBUS2(0xf) -capture_3 asio0 FXBUS2(0x0) -~capture_4 Center EXTOUT(0x11) // mapped to by Center -~capture_5 LFE EXTOUT(0x12) // mapped to by LFE -capture_6 asio3 FXBUS2(0x3) -capture_7 asio4 FXBUS2(0x4) -capture_8 asio5 FXBUS2(0x5) -capture_9 asio6 FXBUS2(0x6) -capture_10 asio7 FXBUS2(0x7) -capture_11 asio8 FXBUS2(0x8) -capture_12 asio9 FXBUS2(0x9) -capture_13 asio10 FXBUS2(0xa) -capture_14 asio11 FXBUS2(0xb) -capture_15 asio12 FXBUS2(0xc) -capture_16 asio13 FXBUS2(0xd) -*/ - -TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt deleted file mode 100644 index de8efbc7e4b..00000000000 --- a/Documentation/sound/alsa/hda_codec.txt +++ /dev/null @@ -1,322 +0,0 @@ -Notes on Universal Interface for Intel High Definition Audio Codec ------------------------------------------------------------------- - -Takashi Iwai <tiwai@suse.de> - - -[Still a draft version] - - -General -======= - -The snd-hda-codec module supports the generic access function for the -High Definition (HD) audio codecs. It's designed to be independent -from the controller code like ac97 codec module. The real accessors -from/to the controller must be implemented in the lowlevel driver. - -The structure of this module is similar with ac97_codec module. -Each codec chip belongs to a bus class which communicates with the -controller. - - -Initialization of Bus Instance -============================== - -The card driver has to create struct hda_bus at first. The template -struct should be filled and passed to the constructor: - -struct hda_bus_template { - void *private_data; - struct pci_dev *pci; - const char *modelname; - struct hda_bus_ops ops; -}; - -The card driver can set and use the private_data field to retrieve its -own data in callback functions. The pci field is used when the patch -needs to check the PCI subsystem IDs, so on. For non-PCI system, it -doesn't have to be set, of course. -The modelname field specifies the board's specific configuration. The -string is passed to the codec parser, and it depends on the parser how -the string is used. -These fields, private_data, pci and modelname are all optional. - -The ops field contains the callback functions as the following: - -struct hda_bus_ops { - int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm); - unsigned int (*get_response)(struct hda_codec *codec); - void (*private_free)(struct hda_bus *); -#ifdef CONFIG_SND_HDA_POWER_SAVE - void (*pm_notify)(struct hda_codec *codec); -#endif -}; - -The command callback is called when the codec module needs to send a -VERB to the controller. It's always a single command. -The get_response callback is called when the codec requires the answer -for the last command. These two callbacks are mandatory and have to -be given. -The third, private_free callback, is optional. It's called in the -destructor to release any necessary data in the lowlevel driver. - -The pm_notify callback is available only with -CONFIG_SND_HDA_POWER_SAVE kconfig. It's called when the codec needs -to power up or may power down. The controller should check the all -belonging codecs on the bus whether they are actually powered off -(check codec->power_on), and optionally the driver may power down the -controller side, too. - -The bus instance is created via snd_hda_bus_new(). You need to pass -the card instance, the template, and the pointer to store the -resultant bus instance. - -int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, - struct hda_bus **busp); - -It returns zero if successful. A negative return value means any -error during creation. - - -Creation of Codec Instance -========================== - -Each codec chip on the board is then created on the BUS instance. -To create a codec instance, call snd_hda_codec_new(). - -int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); - -The first argument is the BUS instance, the second argument is the -address of the codec, and the last one is the pointer to store the -resultant codec instance (can be NULL if not needed). - -The codec is stored in a linked list of bus instance. You can follow -the codec list like: - - struct hda_codec *codec; - list_for_each_entry(codec, &bus->codec_list, list) { - ... - } - -The codec isn't initialized at this stage properly. The -initialization sequence is called when the controls are built later. - - -Codec Access -============ - -To access codec, use snd_hda_codec_read() and snd_hda_codec_write(). -snd_hda_param_read() is for reading parameters. -For writing a sequence of verbs, use snd_hda_sequence_write(). - -There are variants of cached read/write, snd_hda_codec_write_cache(), -snd_hda_sequence_write_cache(). These are used for recording the -register states for the power-management resume. When no PM is needed, -these are equivalent with non-cached version. - -To retrieve the number of sub nodes connected to the given node, use -snd_hda_get_sub_nodes(). The connection list can be obtained via -snd_hda_get_connections() call. - -When an unsolicited event happens, pass the event via -snd_hda_queue_unsol_event() so that the codec routines will process it -later. - - -(Mixer) Controls -================ - -To create mixer controls of all codecs, call -snd_hda_build_controls(). It then builds the mixers and does -initialization stuff on each codec. - - -PCM Stuff -========= - -snd_hda_build_pcms() gives the necessary information to create PCM -streams. When it's called, each codec belonging to the bus stores -codec->num_pcms and codec->pcm_info fields. The num_pcms indicates -the number of elements in pcm_info array. The card driver is supposed -to traverse the codec linked list, read the pcm information in -pcm_info array, and build pcm instances according to them. - -The pcm_info array contains the following record: - -/* PCM information for each substream */ -struct hda_pcm_stream { - unsigned int substreams; /* number of substreams, 0 = not exist */ - unsigned int channels_min; /* min. number of channels */ - unsigned int channels_max; /* max. number of channels */ - hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ - u32 rates; /* supported rates */ - u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ - unsigned int maxbps; /* supported max. bit per sample */ - struct hda_pcm_ops ops; -}; - -/* for PCM creation */ -struct hda_pcm { - char *name; - struct hda_pcm_stream stream[2]; -}; - -The name can be passed to snd_pcm_new(). The stream field contains -the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and -capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver -should pass substreams to snd_pcm_new() for the number of substreams -to create. - -The channels_min, channels_max, rates and formats should be copied to -runtime->hw record. They and maxbps fields are used also to compute -the format value for the HDA codec and controller. Call -snd_hda_calc_stream_format() to get the format value. - -The ops field contains the following callback functions: - -struct hda_pcm_ops { - int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); - int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec, - unsigned int stream_tag, unsigned int format, - struct snd_pcm_substream *substream); - int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec, - struct snd_pcm_substream *substream); -}; - -All are non-NULL, so you can call them safely without NULL check. - -The open callback should be called in PCM open after runtime->hw is -set up. It may override some setting and constraints additionally. -Similarly, the close callback should be called in the PCM close. - -The prepare callback should be called in PCM prepare. This will set -up the codec chip properly for the operation. The cleanup should be -called in hw_free to clean up the configuration. - -The caller should check the return value, at least for open and -prepare callbacks. When a negative value is returned, some error -occurred. - - -Proc Files -========== - -Each codec dumps the widget node information in -/proc/asound/card*/codec#* file. This information would be really -helpful for debugging. Please provide its contents together with the -bug report. - - -Power Management -================ - -It's simple: -Call snd_hda_suspend() in the PM suspend callback. -Call snd_hda_resume() in the PM resume callback. - - -Codec Preset (Patch) -==================== - -To set up and handle the codec functionality fully, each codec may -have a codec preset (patch). It's defined in struct hda_codec_preset: - - struct hda_codec_preset { - unsigned int id; - unsigned int mask; - unsigned int subs; - unsigned int subs_mask; - unsigned int rev; - const char *name; - int (*patch)(struct hda_codec *codec); - }; - -When the codec id and codec subsystem id match with the given id and -subs fields bitwise (with bitmask mask and subs_mask), the callback -patch is called. The patch callback should initialize the codec and -set the codec->patch_ops field. This is defined as below: - - struct hda_codec_ops { - int (*build_controls)(struct hda_codec *codec); - int (*build_pcms)(struct hda_codec *codec); - int (*init)(struct hda_codec *codec); - void (*free)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); - #ifdef CONFIG_PM - int (*suspend)(struct hda_codec *codec, pm_message_t state); - int (*resume)(struct hda_codec *codec); - #endif - #ifdef CONFIG_SND_HDA_POWER_SAVE - int (*check_power_status)(struct hda_codec *codec, - hda_nid_t nid); - #endif - }; - -The build_controls callback is called from snd_hda_build_controls(). -Similarly, the build_pcms callback is called from -snd_hda_build_pcms(). The init callback is called after -build_controls to initialize the hardware. -The free callback is called as a destructor. - -The unsol_event callback is called when an unsolicited event is -received. - -The suspend and resume callbacks are for power management. -They can be NULL if no special sequence is required. When the resume -callback is NULL, the driver calls the init callback and resumes the -registers from the cache. If other handling is needed, you'd need to -write your own resume callback. There, the amp values can be resumed -via - void snd_hda_codec_resume_amp(struct hda_codec *codec); -and the other codec registers via - void snd_hda_codec_resume_cache(struct hda_codec *codec); - -The check_power_status callback is called when the amp value of the -given widget NID is changed. The codec code can turn on/off the power -appropriately from this information. - -Each entry can be NULL if not necessary to be called. - - -Generic Parser -============== - -When the device doesn't match with any given presets, the widgets are -parsed via th generic parser (hda_generic.c). Its support is -limited: no multi-channel support, for example. - - -Digital I/O -=========== - -Call snd_hda_create_spdif_out_ctls() from the patch to create controls -related with SPDIF out. - - -Helper Functions -================ - -snd_hda_get_codec_name() stores the codec name on the given string. - -snd_hda_check_board_config() can be used to obtain the configuration -information matching with the device. Define the model string table -and the table with struct snd_pci_quirk entries (zero-terminated), -and pass it to the function. The function checks the modelname given -as a module parameter, and PCI subsystem IDs. If the matching entry -is found, it returns the config field value. - -snd_hda_add_new_ctls() can be used to create and add control entries. -Pass the zero-terminated array of struct snd_kcontrol_new - -Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be -used for the entry of struct snd_kcontrol_new. - -The input MUX helper callbacks for such a control are provided, too: -snd_hda_input_mux_info() and snd_hda_input_mux_put(). See -patch_realtek.c for example. diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt deleted file mode 100644 index 7a67ff71a9f..00000000000 --- a/Documentation/sound/alsa/hdspm.txt +++ /dev/null @@ -1,362 +0,0 @@ -Software Interface ALSA-DSP MADI Driver - -(translated from German, so no good English ;-), -2004 - winfried ritsch - - - - Full functionality has been added to the driver. Since some of - the Controls and startup-options are ALSA-Standard and only the - special Controls are described and discussed below. - - - hardware functionality: - - - Audio transmission: - - number of channels -- depends on transmission mode - - The number of channels chosen is from 1..Nmax. The reason to - use for a lower number of channels is only resource allocation, - since unused DMA channels are disabled and less memory is - allocated. So also the throughput of the PCI system can be - scaled. (Only important for low performance boards). - - Single Speed -- 1..64 channels - - (Note: Choosing the 56channel mode for transmission or as - receiver, only 56 are transmitted/received over the MADI, but - all 64 channels are available for the mixer, so channel count - for the driver) - - Double Speed -- 1..32 channels - - Note: Choosing the 56-channel mode for - transmission/receive-mode , only 28 are transmitted/received - over the MADI, but all 32 channels are available for the mixer, - so channel count for the driver - - - Quad Speed -- 1..16 channels - - Note: Choosing the 56-channel mode for - transmission/receive-mode , only 14 are transmitted/received - over the MADI, but all 16 channels are available for the mixer, - so channel count for the driver - - Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE) - - Sample Rates -- - - Single Speed -- 32000, 44100, 48000 - - Double Speed -- 64000, 88200, 96000 (untested) - - Quad Speed -- 128000, 176400, 192000 (untested) - - access-mode -- MMAP (memory mapped), Not interleaved - (PCM_NON-INTERLEAVED) - - buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples - - fragments -- 2 - - Hardware-pointer -- 2 Modi - - - The Card supports the readout of the actual Buffer-pointer, - where DMA reads/writes. Since of the bulk mode of PCI it is only - 64 Byte accurate. SO it is not really usable for the - ALSA-mid-level functions (here the buffer-ID gives a better - result), but if MMAP is used by the application. Therefore it - can be configured at load-time with the parameter - precise-pointer. - - - (Hint: Experimenting I found that the pointer is maximum 64 to - large never to small. So if you subtract 64 you always have a - safe pointer for writing, which is used on this mode inside - ALSA. In theory now you can get now a latency as low as 16 - Samples, which is a quarter of the interrupt possibilities.) - - Precise Pointer -- off - interrupt used for pointer-calculation - - Precise Pointer -- on - hardware pointer used. - - Controller: - - - Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to - use the standard mixer-controls, since this would break most of - (especially graphic) ALSA-Mixer GUIs. So Mixer control has be - provided by a 2-dimensional controller using the - hwdep-interface. - - Also all 128+256 Peak and RMS-Meter can be accessed via the - hwdep-interface. Since it could be a performance problem always - copying and converting Peak and RMS-Levels even if you just need - one, I decided to export the hardware structure, so that of - needed some driver-guru can implement a memory-mapping of mixer - or peak-meters over ioctl, or also to do only copying and no - conversion. A test-application shows the usage of the controller. - - Latency Controls --- not implemented !!! - - - Note: Within the windows-driver the latency is accessible of a - control-panel, but buffer-sizes are controlled with ALSA from - hwparams-calls and should not be changed in run-state, I did not - implement it here. - - - System Clock -- suspended !!!! - - Name -- "System Clock Mode" - - Access -- Read Write - - Values -- "Master" "Slave" - - - !!!! This is a hardware-function but is in conflict with the - Clock-source controller, which is a kind of ALSA-standard. I - makes sense to set the card to a special mode (master at some - frequency or slave), since even not using an Audio-application - a studio should have working synchronisations setup. So use - Clock-source-controller instead !!!! - - Clock Source - - Name -- "Sample Clock Source" - - Access -- Read Write - - Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", - "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", - "Internal 96.0 kHz" - - Choose between Master at a specific Frequency and so also the - Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref" - - - !!!! This is no pure hardware function but was implemented by - ALSA by some ALSA-drivers before, so I use it also. !!! - - - Preferred Sync Ref - - Name -- "Preferred Sync Reference" - - Access -- Read Write - - Values -- "Word" "MADI" - - - Within the Auto-sync-Mode the preferred Sync Source can be - chosen. If it is not available another is used if possible. - - Note: Since MADI has a much higher bit-rate than word-clock, the - card should synchronise better in MADI Mode. But since the - RME-PLL is very good, there are almost no problems with - word-clock too. I never found a difference. - - - TX 64 channel --- - - Name -- "TX 64 channels mode" - - Access -- Read Write - - Values -- 0 1 - - Using 64-channel-modus (1) or 56-channel-modus for - MADI-transmission (0). - - - Note: This control is for output only. Input-mode is detected - automatically from hardware sending MADI. - - - Clear TMS --- - - Name -- "Clear Track Marker" - - Access -- Read Write - - Values -- 0 1 - - - Don't use to lower 5 Audio-bits on AES as additional Bits. - - - Safe Mode oder Auto Input --- - - Name -- "Safe Mode" - - Access -- Read Write - - Values -- 0 1 - - (default on) - - If on (1), then if either the optical or coaxial connection - has a failure, there is a takeover to the working one, with no - sample failure. Its only useful if you use the second as a - backup connection. - - Input --- - - Name -- "Input Select" - - Access -- Read Write - - Values -- optical coaxial - - - Choosing the Input, optical or coaxial. If Safe-mode is active, - this is the preferred Input. - --------------- Mixer ---------------------- - - Mixer - - Name -- "Mixer" - - Access -- Read Write - - Values - <channel-number 0-127> <Value 0-65535> - - - Here as a first value the channel-index is taken to get/set the - corresponding mixer channel, where 0-63 are the input to output - fader and 64-127 the playback to outputs fader. Value 0 - is channel muted 0 and 32768 an amplification of 1. - - Chn 1-64 - - fast mixer for the ALSA-mixer utils. The diagonal of the - mixer-matrix is implemented from playback to output. - - - Line Out - - Name -- "Line Out" - - Access -- Read Write - - Values -- 0 1 - - Switching on and off the analog out, which has nothing to do - with mixing or routing. the analog outs reflects channel 63,64. - - ---- information (only read access): - - Sample Rate - - Name -- "System Sample Rate" - - Access -- Read-only - - getting the sample rate. - - - External Rate measured - - Name -- "External Rate" - - Access -- Read only - - - Should be "Autosync Rate", but Name used is - ALSA-Scheme. External Sample frequency liked used on Autosync is - reported. - - - MADI Sync Status - - Name -- "MADI Sync Lock Status" - - Access -- Read - - Values -- 0,1,2 - - MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced. - - - Word Clock Sync Status - - Name -- "Word Clock Lock Status" - - Access -- Read - - Values -- 0,1,2 - - Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced. - - AutoSync - - Name -- "AutoSync Reference" - - Access -- Read - - Values -- "WordClock", "MADI", "None" - - Sync-Reference is either "WordClock", "MADI" or none. - - RX 64ch --- noch nicht implementiert - - MADI-Receiver is in 64 channel mode oder 56 channel mode. - - - AB_inp --- not tested - - Used input for Auto-Input. - - - actual Buffer Position --- not implemented - - !!! this is a ALSA internal function, so no control is used !!! - - - -Calling Parameter: - - index int array (min = 1, max = 8), - "Index value for RME HDSPM interface." card-index within ALSA - - note: ALSA-standard - - id string array (min = 1, max = 8), - "ID string for RME HDSPM interface." - - note: ALSA-standard - - enable int array (min = 1, max = 8), - "Enable/disable specific HDSPM sound-cards." - - note: ALSA-standard - - precise_ptr int array (min = 1, max = 8), - "Enable precise pointer, or disable." - - note: Use only when the application supports this (which is a special case). - - line_outs_monitor int array (min = 1, max = 8), - "Send playback streams to analog outs by default." - - - note: each playback channel is mixed to the same numbered output - channel (routed). This is against the ALSA-convention, where all - channels have to be muted on after loading the driver, but was - used before on other cards, so i historically use it again) - - - - enable_monitor int array (min = 1, max = 8), - "Enable Analog Out on Channel 63/64 by default." - - note: here the analog output is enabled (but not routed).
\ No newline at end of file diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt deleted file mode 100644 index 9657e809922..00000000000 --- a/Documentation/sound/alsa/powersave.txt +++ /dev/null @@ -1,41 +0,0 @@ -Notes on Power-Saving Mode -========================== - -AC97 and HD-audio drivers have the automatic power-saving mode. -This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE -and CONFIG_SND_HDA_POWER_SAVE options, respectively. - -With the automatic power-saving, the driver turns off the codec power -appropriately when no operation is required. When no applications use -the device and/or no analog loopback is set, the power disablement is -done fully or partially. It'll save a certain power consumption, thus -good for laptops (even for desktops). - -The time-out for automatic power-off can be specified via power_save -module option of snd-ac97-codec and snd-hda-intel modules. Specify -the time-out value in seconds. 0 means to disable the automatic -power-saving. The default value of timeout is given via -CONFIG_SND_AC97_POWER_SAVE_DEFAULT and -CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 -(the minimum value) isn't recommended because many applications try to -reopen the device frequently. 10 would be a good choice for normal -operations. - -The power_save option is exported as writable. This means you can -adjust the value via sysfs on the fly. For example, to turn on the -automatic power-save mode with 10 seconds, write to -/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): - - # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save - - -Note that you might hear click noise/pop when changing the power -state. Also, it often takes certain time to wake up from the -power-down to the active state. These are often hardly to fix, so -don't report extra bug reports unless you have a fix patch ;-) - -For HD-audio interface, there is another module option, -power_save_controller. This enables/disables the power-save mode of -the controller side. Setting this on may reduce a bit more power -consumption, but might result in longer wake-up time and click noise. -Try to turn it off when you experience such a thing too often. diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html deleted file mode 100644 index d9776cf60c0..00000000000 --- a/Documentation/sound/alsa/seq_oss.html +++ /dev/null @@ -1,409 +0,0 @@ -<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> -<HTML> -<HEAD> - <TITLE>OSS Sequencer Emulation on ALSA</TITLE> -</HEAD> -<BODY> - -<CENTER> -<H1> - -<HR WIDTH="100%"></H1></CENTER> - -<CENTER> -<H1> -OSS Sequencer Emulation on ALSA</H1></CENTER> - -<HR WIDTH="100%"> -<P>Copyright (c) 1998,1999 by Takashi Iwai -<TT><A HREF="mailto:iwai@ww.uni-erlangen.de"><iwai@ww.uni-erlangen.de></A></TT> -<P>ver.0.1.8; Nov. 16, 1999 -<H2> - -<HR WIDTH="100%"></H2> - -<H2> -1. Description</H2> -This directory contains the OSS sequencer emulation driver on ALSA. Note -that this program is still in the development state. -<P>What this does - it provides the emulation of the OSS sequencer, access -via -<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices. -The most of applications using OSS can run if the appropriate ALSA -sequencer is prepared. -<P>The following features are emulated by this driver: -<UL> -<LI> -Normal sequencer and MIDI events:</LI> - -<BR>They are converted to the ALSA sequencer events, and sent to the corresponding -port. -<LI> -Timer events:</LI> - -<BR>The timer is not selectable by ioctl. The control rate is fixed to -100 regardless of HZ. That is, even on Alpha system, a tick is always -1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>. - -<LI> -Patch loading:</LI> - -<BR>It purely depends on the synth drivers whether it's supported since -the patch loading is realized by callback to the synth driver. -<LI> -I/O controls:</LI> - -<BR>Most of controls are accepted. Some controls -are dependent on the synth driver, as well as even on original OSS.</UL> -Furthermore, you can find the following advanced features: -<UL> -<LI> -Better queue mechanism:</LI> - -<BR>The events are queued before processing them. -<LI> -Multiple applications:</LI> - -<BR>You can run two or more applications simultaneously (even for OSS sequencer)! -However, each MIDI device is exclusive - that is, if a MIDI device is opened -once by some application, other applications can't use it. No such a restriction -in synth devices. -<LI> -Real-time event processing:</LI> - -<BR>The events can be processed in real time without using out of bound -ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed -events will be processed in real-time without queued. To switch off the -real-time mode, send RELTIME 0 event. -<LI> -<TT>/proc</TT> interface:</LI> - -<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT> -at any time. In the later version, configuration will be changed via <TT>/proc</TT> -interface, too.</UL> - -<H2> -2. Installation</H2> -Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>) -and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT> -will be created. If the synth module of your sound card supports for OSS -emulation (so far, only Emu8000 driver), this module will be loaded automatically. -Otherwise, you need to load this module manually. -<P>At beginning, this module probes all the MIDI ports which have been -already connected to the sequencer. Once after that, the creation and deletion -of ports are watched by announcement mechanism of ALSA sequencer. -<P>The available synth and MIDI devices can be found in proc interface. -Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example, -if you use an AWE64 card, you'll see like the following: -<PRE> OSS sequencer emulation version 0.1.8 - ALSA client number 63 - ALSA receiver port 0 - - Number of applications: 0 - - Number of synth devices: 1 - - synth 0: [EMU8000] - type 0x1 : subtype 0x20 : voices 32 - capabilties : ioctl enabled / load_patch enabled - - Number of MIDI devices: 3 - - midi 0: [Emu8000 Port-0] ALSA port 65:0 - capability write / opened none - - midi 1: [Emu8000 Port-1] ALSA port 65:1 - capability write / opened none - - midi 2: [0: MPU-401 (UART)] ALSA port 64:0 - capability read/write / opened none</PRE> -Note that the device number may be different from the information of -<TT>/proc/asound/oss-devices</TT> -or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT> -to play via OSS sequencer emulation. -<H2> -3. Using Synthesizer Devices</H2> -Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 -and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload, -too. -<P>If the lowlevel driver supports multiple access to synth devices (like -Emu8000 driver), two or more applications are allowed to run at the same -time. -<H2> -4. Using MIDI Devices</H2> -So far, only MIDI output was tested. MIDI input was not checked at all, -but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>. -Be aware that these numbers are mostly different from the list in -<TT>/proc/asound/oss-devices</TT>. -<H2> -5. Module Options</H2> -The following module options are available: -<UL> -<LI> -<TT>maxqlen</TT></LI> - -<BR>specifies the maximum read/write queue length. This queue is private -for OSS sequencer, so that it is independent from the queue length of ALSA -sequencer. Default value is 1024. -<LI> -<TT>seq_oss_debug</TT></LI> - -<BR>specifies the debug level and accepts zero (= no debug message) or -positive integer. Default value is 0.</UL> - -<H2> -6. Queue Mechanism</H2> -OSS sequencer emulation uses an ALSA priority queue. The -events from <TT>/dev/sequencer</TT> are processed and put onto the queue -specified by module option. -<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning. -The timing events are also parsed at this moment, so that the events may -be processed in real-time. Sending an event ABSTIME 0 switches the operation -mode to real-time mode, and sending an event RELTIME 0 switches it off. -In the real-time mode, all events are dispatched immediately. -<P>The queued events are dispatched to the corresponding ALSA sequencer -ports after scheduled time by ALSA sequencer dispatcher. -<P>If the write-queue is full, the application sleeps until a certain amount -(as default one half) becomes empty in blocking mode. The synchronization -to write timing was implemented, too. -<P>The input from MIDI devices or echo-back events are stored on read FIFO -queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the -process will be awaked. - -<H2> -7. Interface to Synthesizer Device</H2> - -<H3> -7.1. Registration</H3> -To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT> -function. -<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices, - snd_seq_oss_callback_t *oper, void *private_data)</PRE> -The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and -<TT>nvoices</TT> -are used for making the appropriate synth_info structure for ioctl. The -return value is an index number of this device. This index must be remembered -for unregister. If registration is failed, -errno will be returned. -<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>: -<PRE>int snd_seq_oss_synth_unregister(int index),</PRE> -where the <TT>index</TT> is the index number returned by register function. -<H3> -7.2. Callbacks</H3> -OSS synthesizer devices have capability for sample downloading and ioctls -like sample reset. In OSS emulation, these special features are realized -by using callbacks. The registration argument oper is used to specify these -callbacks. The following callback functions must be defined: -<PRE>snd_seq_oss_callback_t: - int (*open)(snd_seq_oss_arg_t *p, void *closure); - int (*close)(snd_seq_oss_arg_t *p); - int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg); - int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count); - int (*reset)(snd_seq_oss_arg_t *p); -Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed -to be NULL. -<P>Each callback function takes the argument type snd_seq_oss_arg_t as the -first argument. -<PRE>struct snd_seq_oss_arg_t { - int app_index; - int file_mode; - int seq_mode; - snd_seq_addr_t addr; - void *private_data; - int event_passing; -};</PRE> -The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and -<TT>seq_mode</TT> -are initialized by OSS sequencer. The <TT>app_index</TT> is the application -index which is unique to each application opening OSS sequencer. The -<TT>file_mode</TT> -is bit-flags indicating the file operation mode. See -<TT>seq_oss.h</TT> -for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In -the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used. -<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be -filled by the synth driver at open callback. The <TT>addr</TT> contains -the address of ALSA sequencer port which is assigned to this device. If -the driver allocates memory for <TT>private_data</TT>, it must be released -in close callback by itself. -<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on -/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded -as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT> -mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT> -mode checks the note above 128 and regards it as key pressure event (mainly -for Emu8000 driver). -<H4> -7.2.1. Open Callback</H4> -The <TT>open</TT> is called at each time this device is opened by an application -using OSS sequencer. This must not be NULL. Typically, the open callback -does the following procedure: -<OL> -<LI> -Allocate private data record.</LI> - -<LI> -Create an ALSA sequencer port.</LI> - -<LI> -Set the new port address on arg->addr.</LI> - -<LI> -Set the private data record pointer on arg->private_data.</LI> -</OL> -Note that the type bit-flags in port_info of this synth port must NOT contain -<TT>TYPE_MIDI_GENERIC</TT> -bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT> -bit should NOT be included, too. This is necessary to tell it from other -normal MIDI devices. If the open procedure succeeded, return zero. Otherwise, -return -errno. -<H4> -7.2.2 Ioctl Callback</H4> -The <TT>ioctl</TT> callback is called when the sequencer receives device-specific -ioctls. The following two ioctls should be processed by this callback: -<UL> -<LI> -<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI> - -<BR>reset all samples on memory -- return 0 -<LI> -<TT>IOCTL_SYNTH_MEMAVL</TT></LI> - -<BR>return the available memory size -<LI> -<TT>FM_4OP_ENABLE</TT></LI> - -<BR>can be ignored usually</UL> -The other ioctls are processed inside the sequencer without passing to -the lowlevel driver. -<H4> -7.2.3 Load_Patch Callback</H4> -The <TT>load_patch</TT> callback is used for sample-downloading. This callback -must read the data on user-space and transfer to each device. Return 0 -if succeeded, and -errno if failed. The format argument is the patch key -in patch_info record. The buf is user-space pointer where patch_info record -is stored. The offs can be ignored. The count is total data size of this -sample data. -<H4> -7.2.4 Close Callback</H4> -The <TT>close</TT> callback is called when this device is closed by the -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. -<H4> -7.2.5 Reset Callback</H4> -The <TT>reset</TT> callback is called when sequencer device is reset or -closed by applications. The callback should turn off the sounds on the -relevant port immediately, and initialize the status of the port. If this -callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the -port. -<H3> -7.3 Events</H3> -Most of the events are processed by sequencer and translated to the adequate -ALSA sequencer events, so that each synth device can receive by input_event -callback of ALSA sequencer port. The following ALSA events should be implemented -by the driver: -<BR> -<TABLE BORDER WIDTH="75%" NOSAVE > -<TR NOSAVE> -<TD NOSAVE><B>ALSA event</B></TD> - -<TD><B>Original OSS events</B></TD> -</TR> - -<TR> -<TD>NOTEON</TD> - -<TD>SEQ_NOTEON -<BR>MIDI_NOTEON</TD> -</TR> - -<TR> -<TD>NOTE</TD> - -<TD>SEQ_NOTEOFF -<BR>MIDI_NOTEOFF</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>KEYPRESS</TD> - -<TD>MIDI_KEY_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD>CHANPRESS</TD> - -<TD NOSAVE>SEQ_AFTERTOUCH -<BR>MIDI_CHN_PRESSURE</TD> -</TR> - -<TR NOSAVE> -<TD NOSAVE>PGMCHANGE</TD> - -<TD NOSAVE>SEQ_PGMCHANGE -<BR>MIDI_PGM_CHANGE</TD> -</TR> - -<TR> -<TD>PITCHBEND</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER) -<BR>MIDI_PITCH_BEND</TD> -</TR> - -<TR> -<TD>CONTROLLER</TD> - -<TD>MIDI_CTL_CHANGE -<BR>SEQ_BALANCE (with CTL_PAN)</TD> -</TR> - -<TR> -<TD>CONTROL14</TD> - -<TD>SEQ_CONTROLLER</TD> -</TR> - -<TR> -<TD>REGPARAM</TD> - -<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD> -</TR> - -<TR> -<TD>SYSEX</TD> - -<TD>SEQ_SYSEX</TD> -</TR> -</TABLE> - -<P>The most of these behavior can be realized by MIDI emulation driver -included in the Emu8000 lowlevel driver. In the future release, this module -will be independent. -<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event -type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte -packets without any modification. The lowlevel driver should process these -events appropriately. -<H2> -8. Interface to MIDI Device</H2> -Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer -ports automatically by receiving announcement from ALSA sequencer, the -MIDI devices don't need to be registered explicitly like synth devices. -However, the MIDI port_info registered to ALSA sequencer must include a group -name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or -<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>, -must be defined, too. If these conditions are not satisfied, the port is not -registered as OSS sequencer MIDI device. -<P>The events via MIDI devices are parsed in OSS sequencer and converted -to the corresponding ALSA sequencer events. The input from MIDI sequencer -is also converted to MIDI byte events by OSS sequencer. This works just -a reverse way of seq_midi module. -<H2> -9. Known Problems / TODO's</H2> - -<UL> -<LI> -Patch loading via ALSA instrument layer is not implemented yet.</LI> -</UL> - -</BODY> -</HTML> diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt deleted file mode 100644 index c1919559d50..00000000000 --- a/Documentation/sound/alsa/serial-u16550.txt +++ /dev/null @@ -1,88 +0,0 @@ - - Serial UART 16450/16550 MIDI driver - =================================== - -The adaptor module parameter allows you to select either: - - 0 - Roland Soundcanvas support (default) - 1 - Midiator MS-124T support (1) - 2 - Midiator MS-124W S/A mode (2) - 3 - MS-124W M/B mode support (3) - 4 - Generic device with multiple input support (4) - -For the Midiator MS-124W, you must set the physical M-S and A-B -switches on the Midiator to match the driver mode you select. - -In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported -(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver -sends the nonstandard MIDI command sequence F5 NN, where NN is the substream -number plus 1. Roland modules use this command to switch between different -"parts", so this feature lets you treat each part as a distinct raw MIDI -substream. The driver provides no way to send F5 00 (no selection) or to not -send the F5 NN command sequence at all; perhaps it ought to. - -Usage example for simple serial converter: - - /sbin/setserial /dev/ttyS0 uart none - /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200 - -Usage example for Roland SoundCanvas with 4 MIDI ports: - - /sbin/setserial /dev/ttyS0 uart none - /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4 - -In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs -module parameter is automatically set to 1. The driver sends the same data to -all four MIDI Out connectors. Set the A-B switch and the speed module -parameter to match (A=19200, B=9600). - -Usage example for MS-124T, with A-B switch in A position: - - /sbin/setserial /dev/ttyS0 uart none - /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \ - speed=19200 - -In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0); -the outs module parameter is automatically set to 1. The driver sends -the same data to all four MIDI Out connectors at full MIDI speed. - -Usage example for S/A mode: - - /sbin/setserial /dev/ttyS0 uart none - /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2 - -In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams; -the outs module parameter is automatically set to 16. The substream -number gives a bitmask of which MIDI Out connectors the data should be -sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to -Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports. -As a special case, midiCnD0 also sends to all ports, since it is not useful -to send the data to no ports. M/B mode has extra overhead to select the MIDI -Out for each byte, so the aggregate data rate across all four MIDI Outs is -at most one byte every 520 us, as compared with the full MIDI data rate of -one byte every 320 us per port. - -Usage example for M/B mode: - - /sbin/setserial /dev/ttyS0 uart none - /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3 - -The MS-124W hardware's M/A mode is currently not supported. This mode allows -the MIDI Outs to act independently at double the aggregate throughput of M/B, -but does not allow sending the same byte simultaneously to multiple MIDI Outs. -The M/A protocol requires the driver to twiddle the modem control lines under -timing constraints, so it would be a bit more complicated to implement than -the other modes. - -Midiator models other than MS-124W and MS-124T are currently not supported. -Note that the suffix letter is significant; the MS-124 and MS-124B are not -compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114. -I do have documentation (tim.mann@compaq.com) that partially covers these models, -but no units to experiment with. The MS-124W support is tested with a real unit. -The MS-124T support is untested, but should work. - -The Generic driver supports multiple input and output substreams over a single -serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the -appropriate input or output stream (depending on the data direction). -Additionally, the CTS signal is used to regulate the data flow. The number of -inputs is specified by the ins parameter. diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt deleted file mode 100644 index c9679264c55..00000000000 --- a/Documentation/sound/alsa/soc/DAI.txt +++ /dev/null @@ -1,56 +0,0 @@ -ASoC currently supports the three main Digital Audio Interfaces (DAI) found on -SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. - - -AC97 -==== - - AC97 is a five wire interface commonly found on many PC sound cards. It is -now also popular in many portable devices. This DAI has a reset line and time -multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. -The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the -frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 -frame is 21uS long and is divided into 13 time slots. - -The AC97 specification can be found at :- -http://www.intel.com/p/en_US/business/design - - -I2S -=== - - I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and -Rx lines are used for audio transmission, whilst the bit clock (BCLK) and -left/right clock (LRC) synchronise the link. I2S is flexible in that either the -controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock -usually varies depending on the sample rate and the master system clock -(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate -ADC and DAC LRCLKs, this allows for simultaneous capture and playback at -different sample rates. - -I2S has several different operating modes:- - - o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC - transition. - - o Left Justified - MSB is transmitted on transition of LRC. - - o Right Justified - MSB is transmitted sample size BCLKs before LRC - transition. - -PCM -=== - -PCM is another 4 wire interface, very similar to I2S, which can support a more -flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used -to synchronise the link whilst the Tx and Rx lines are used to transmit and -receive the audio data. Bit clock usually varies depending on sample rate -whilst sync runs at the sample rate. PCM also supports Time Division -Multiplexing (TDM) in that several devices can use the bus simultaneously (this -is sometimes referred to as network mode). - -Common PCM operating modes:- - - o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. - - o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt deleted file mode 100644 index b1300162e01..00000000000 --- a/Documentation/sound/alsa/soc/clocking.txt +++ /dev/null @@ -1,51 +0,0 @@ -Audio Clocking -============== - -This text describes the audio clocking terms in ASoC and digital audio in -general. Note: Audio clocking can be complex! - - -Master Clock ------------- - -Every audio subsystem is driven by a master clock (sometimes referred to as MCLK -or SYSCLK). This audio master clock can be derived from a number of sources -(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct -audio playback and capture sample rates. - -Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that -their speed can be altered by software (depending on the system use and to save -power). Other master clocks are fixed at a set frequency (i.e. crystals). - - -DAI Clocks ----------- -The Digital Audio Interface is usually driven by a Bit Clock (often referred to -as BCLK). This clock is used to drive the digital audio data across the link -between the codec and CPU. - -The DAI also has a frame clock to signal the start of each audio frame. This -clock is sometimes referred to as LRC (left right clock) or FRAME. This clock -runs at exactly the sample rate (LRC = Rate). - -Bit Clock can be generated as follows:- - -BCLK = MCLK / x - - or - -BCLK = LRC * x - - or - -BCLK = LRC * Channels * Word Size - -This relationship depends on the codec or SoC CPU in particular. In general -it is best to configure BCLK to the lowest possible speed (depending on your -rate, number of channels and word size) to save on power. - -It is also desirable to use the codec (if possible) to drive (or master) the -audio clocks as it usually gives more accurate sample rates than the CPU. - - - diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt deleted file mode 100644 index bce23a4a787..00000000000 --- a/Documentation/sound/alsa/soc/codec.txt +++ /dev/null @@ -1,195 +0,0 @@ -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 deleted file mode 100644 index 05bf5a0eee4..00000000000 --- a/Documentation/sound/alsa/soc/dapm.txt +++ /dev/null @@ -1,294 +0,0 @@ -Dynamic Audio Power Management for Portable Devices -=================================================== - -1. Description -============== - -Dynamic Audio Power Management (DAPM) is designed to allow portable -Linux devices to use the minimum amount of power within the audio -subsystem at all times. It is independent of other kernel PM and as -such, can easily co-exist with the other PM systems. - -DAPM is also completely transparent to all user space applications as -all power switching is done within the ASoC core. No code changes or -recompiling are required for user space applications. DAPM makes power -switching decisions based upon any audio stream (capture/playback) -activity and audio mixer settings within the device. - -DAPM spans the whole machine. It covers power control within the entire -audio subsystem, this includes internal codec power blocks and machine -level power systems. - -There are 4 power domains within DAPM - - 1. Codec 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 deleted file mode 100644 index fcf82a41729..00000000000 --- a/Documentation/sound/alsa/soc/jack.txt +++ /dev/null @@ -1,71 +0,0 @@ -ASoC jack detection -=================== - -ALSA has a standard API for representing physical jacks to user space, -the kernel side of which can be seen in include/sound/jack.h. ASoC -provides a version of this API adding two additional features: - - - It allows more than one jack detection method to work together on one - user visible jack. In embedded systems it is common for multiple - to be present on a single jack but handled by separate bits of - hardware. - - - Integration with DAPM, allowing DAPM endpoints to be updated - automatically based on the detected jack status (eg, turning off the - headphone outputs if no headphones are present). - -This is done by splitting the jacks up into three things working -together: the jack itself represented by a struct snd_soc_jack, sets of -snd_soc_jack_pins representing DAPM endpoints to update and blocks of -code providing jack reporting mechanisms. - -For example, a system may have a stereo headset jack with two reporting -mechanisms, one for the headphone and one for the microphone. Some -systems won't be able to use their speaker output while a headphone is -connected and so will want to make sure to update both speaker and -headphone when the headphone jack status changes. - -The jack - struct snd_soc_jack -============================== - -This represents a physical jack on the system and is what is visible to -user space. The jack itself is completely passive, it is set up by the -machine driver and updated by jack detection methods. - -Jacks are created by the machine driver calling snd_soc_jack_new(). - -snd_soc_jack_pin -================ - -These represent a DAPM pin to update depending on some of the status -bits supported by the jack. Each snd_soc_jack has zero or more of these -which are updated automatically. They are created by the machine driver -and associated with the jack using snd_soc_jack_add_pins(). The status -of the endpoint may configured to be the opposite of the jack status if -required (eg, enabling a built in microphone if a microphone is not -connected via a jack). - -Jack detection methods -====================== - -Actual jack detection is done by code which is able to monitor some -input to the system and update a jack by calling snd_soc_jack_report(), -specifying a subset of bits to update. The jack detection code should -be set up by the machine driver, taking configuration for the jack to -update and the set of things to report when the jack is connected. - -Often this is done based on the status of a GPIO - a handler for this is -provided by the snd_soc_jack_add_gpio() function. Other methods are -also available, for example integrated into CODECs. One example of -CODEC integrated jack detection can be see in the WM8350 driver. - -Each jack may have multiple reporting mechanisms, though it will need at -least one to be useful. - -Machine drivers -=============== - -These are all hooked together by the machine driver depending on the -system hardware. The machine driver will set up the snd_soc_jack and -the list of pins to update then set up one or more jack detection -mechanisms to update that jack based on their current status. diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt deleted file mode 100644 index d50c14df341..00000000000 --- a/Documentation/sound/alsa/soc/machine.txt +++ /dev/null @@ -1,91 +0,0 @@ -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. - -struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. - -/* corgi digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link corgi_dai = { - .name = "WM8731", - .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", - .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", - .codec_name = "wm8713-codec.0-001a", - .init = corgi_wm8731_init, - .ops = &corgi_ops, -}; - -struct snd_soc_card then sets up the machine with its DAIs. e.g. - -/* corgi audio machine driver */ -static struct snd_soc_card snd_soc_corgi = { - .name = "Corgi", - .dai_link = &corgi_dai, - .num_links = 1, -}; - - -Machine Power Map ------------------ - -The machine driver can optionally extend the codec power map and to become an -audio power map of the audio subsystem. This allows for automatic power up/down -of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack -sockets in the machine init function. - - -Machine Controls ----------------- - -Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt deleted file mode 100644 index 138ac88c146..00000000000 --- a/Documentation/sound/alsa/soc/overview.txt +++ /dev/null @@ -1,86 +0,0 @@ -ALSA SoC Layer -============== - -The overall project goal of the ALSA System on Chip (ASoC) layer is to -provide better ALSA support for embedded system-on-chip processors (e.g. -pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC -subsystem there was some support in the kernel for SoC audio, however it -had some limitations:- - - * Codec drivers were often tightly coupled to the underlying SoC - CPU. This is not ideal and leads to code duplication - for example, - Linux had different wm8731 drivers for 4 different SoC platforms. - - * There was no standard method to signal user initiated audio events (e.g. - Headphone/Mic insertion, Headphone/Mic detection after an insertion - event). These are quite common events on portable devices and often require - machine specific code to re-route audio, enable amps, etc., after such an - event. - - * Drivers tended to power up the entire codec when playing (or - recording) audio. This is fine for a PC, but tends to waste a lot of - power on portable devices. There was also no support for saving - power via changing codec oversampling rates, bias currents, etc. - - -ASoC Design -=========== - -The ASoC layer is designed to address these issues and provide the following -features :- - - * Codec independence. Allows reuse of codec drivers on other platforms - and machines. - - * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC - interface and codec registers its audio interface capabilities with the - core and are subsequently matched and configured when the application - hardware parameters are known. - - * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to - its minimum power state at all times. This includes powering up/down - internal power blocks depending on the internal codec audio routing and any - active streams. - - * Pop and click reduction. Pops and clicks can be reduced by powering the - codec up/down in the correct sequence (including using digital mute). ASoC - signals the codec when to change power states. - - * Machine specific controls: Allow machines to add controls to the sound card - (e.g. volume control for speaker amplifier). - -To achieve all this, ASoC basically splits an embedded audio system into 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 deleted file mode 100644 index d57efad37e0..00000000000 --- a/Documentation/sound/alsa/soc/platform.txt +++ /dev/null @@ -1,66 +0,0 @@ -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 deleted file mode 100644 index e1e74daa449..00000000000 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ /dev/null @@ -1,52 +0,0 @@ -Audio Pops and Clicks -===================== - -Pops and clicks are unwanted audio artifacts caused by the powering up and down -of components within the audio subsystem. This is noticeable on PCs when an -audio module is either loaded or unloaded (at module load time the sound card is -powered up and causes a popping noise on the speakers). - -Pops and clicks can be more frequent on portable systems with DAPM. This is -because the components within the subsystem are being dynamically powered -depending on the audio usage and this can subsequently cause a small pop or -click every time a component power state is changed. - - -Minimising Playback Pops and Clicks -=================================== - -Playback pops in portable audio subsystems cannot be completely eliminated -currently, however future audio codec hardware will have better pop and click -suppression. Pops can be reduced within playback by powering the audio -components in a specific order. This order is different for startup and -shutdown and follows some basic rules:- - - Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute - - Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC - -This assumes that the codec PCM output path from the DAC is via a mixer and then -a PGA (programmable gain amplifier) before being output to the speakers. - - -Minimising Capture Pops and Clicks -================================== - -Capture artifacts are somewhat easier to get rid as we can delay activating the -ADC until all the pops have occurred. This follows similar power rules to -playback in that components are powered in a sequence depending upon stream -startup or shutdown. - - Startup Order - Input PGA --> Mixers --> ADC - - Shutdown Order - ADC --> Mixers --> Input PGA - - -Zipper Noise -============ -An unwanted zipper noise can occur within the audio playback or capture stream -when a volume control is changed near its maximum gain value. The zipper noise -is heard when the gain increase or decrease changes the mean audio signal -amplitude too quickly. It can be minimised by enabling the zero cross setting -for each volume control. The ZC forces the gain change to occur when the signal -crosses the zero amplitude line. diff --git a/Documentation/sound/oss/ALS b/Documentation/sound/oss/ALS deleted file mode 100644 index bf10bed4574..00000000000 --- a/Documentation/sound/oss/ALS +++ /dev/null @@ -1,66 +0,0 @@ -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@just42.net -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@just42.net - - updated information for kernel 2.4.x. diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16 deleted file mode 100644 index ea8549faede..00000000000 --- a/Documentation/sound/oss/AudioExcelDSP16 +++ /dev/null @@ -1,101 +0,0 @@ -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) - -A configuration file in /etc/modprobe.d/ directory 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 configuration -files under /etc/modprobe.d/ directory: - -softdep ad1848 pre: aedsp16 -softdep opl3 pre: aedsp16 - -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 -<ctrl>-<alt>-<del> 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) <ctrl>-<alt>-<del> 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 <sjg95@unixfe.rl.ac.uk> -> 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 deleted file mode 100644 index 8a5fd1611c6..00000000000 --- a/Documentation/sound/oss/CMI8330 +++ /dev/null @@ -1,152 +0,0 @@ -Documentation for CMI 8330 (SoundPRO) -------------------------------------- -Alessandro Zummo <azummo@ita.flashnet.it> - -( 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 <azummo@ita.flashnet.it> -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 <Stefan.Laudat@asit.ro> - -[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: - -<snip below> - - -(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) - -<end of snip> - - 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 <elysian@ethereal.torsion.org> suggests the following in -a /etc/modprobe.d/*conf file: - -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 deleted file mode 100644 index bba93b4d2de..00000000000 --- a/Documentation/sound/oss/ESS +++ /dev/null @@ -1,34 +0,0 @@ -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 deleted file mode 100644 index 55e922f21bc..00000000000 --- a/Documentation/sound/oss/ESS1868 +++ /dev/null @@ -1,55 +0,0 @@ -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 deleted file mode 100644 index 42da2d8fa37..00000000000 --- a/Documentation/sound/oss/Introduction +++ /dev/null @@ -1,459 +0,0 @@ -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 <stodolsk@erols.com>]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.d/oss.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.d/*.conf -3. move the sound modules from /lib/modules/<kernel>/misc - (for example, I make a /lib/modules/<kernel>/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.d/*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 <program> 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.d/*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 deleted file mode 100644 index e4a18bb7f73..00000000000 --- a/Documentation/sound/oss/MultiSound +++ /dev/null @@ -1,1137 +0,0 @@ -#! /bin/sh -# -# Turtle Beach MultiSound Driver Notes -# -- Andrew Veliath <andrewtv@usa.net> -# -# Last update: September 10, 1998 -# Corresponding msnd driver: 0.8.3 -# -# ** This file is a README (top part) and shell archive (bottom part). -# The corresponding archived utility sources can be unpacked by -# running `sh MultiSound' (the utilities are only needed for the -# Pinnacle and Fiji cards). ** -# -# -# -=-=- Getting Firmware -=-=- -# ~~~~~~~~~~~~~~~~~~~~~~~~~~~~ -# -# See the section `Obtaining and Creating Firmware Files' in this -# document for instructions on obtaining the necessary firmware -# files. -# -# -# Supported Features -# ~~~~~~~~~~~~~~~~~~ -# -# Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is -# not currently available) and mixer functionality (/dev/mixer) are -# supported (memory mapped digital audio is not yet supported). -# Digital transfers and monitoring can be done as well if you have -# the digital daughterboard (see the section on using the S/PDIF port -# for more information). -# -# Support for the Turtle Beach MultiSound Hurricane architecture is -# composed of the following modules (these can also operate compiled -# into the kernel): -# -# 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 <andrewtv@ztransform.velsoft.com>. -# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'. -# -# Existing files will *not* be overwritten unless `-c' is specified. -# -# This shar contains: -# length mode name -# ------ ---------- ------------------------------------------ -# 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 <stdio.h> -#include <unistd.h> -#include <fcntl.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <sys/ioctl.h> -#include <sys/soundcard.h> -X -int main(int argc, char *argv[]) -{ -X int fd; -X unsigned long recmask, recsrc; -X -X if (argc != 2) { -X fprintf(stderr, "usage: setdigital <mixer device>\n"); -X exit(1); -X } -X -X if ((fd = open(argv[1], O_RDWR)) < 0) { -X perror(argv[1]); -X exit(1); -X } -X -X if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) { -X fprintf(stderr, "error: ioctl read recording mask failed\n"); -X perror("ioctl"); -X close(fd); -X exit(1); -X } -X -X if (!(recmask & SOUND_MASK_DIGITAL1)) { -X fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n"); -X close(fd); -X exit(1); -X } -X -X if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) { -X fprintf(stderr, "error: ioctl read recording source failed\n"); -X perror("ioctl"); -X close(fd); -X exit(1); -X } -X -X recsrc |= SOUND_MASK_DIGITAL1; -X -X if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) { -X fprintf(stderr, "error: ioctl write recording source failed\n"); -X perror("ioctl"); -X close(fd); -X exit(1); -X } -X -X close(fd); -X -X return 0; -} -SHAR_EOF - $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' && - chmod 0664 'MultiSound.d/setdigital.c' || - $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed' - if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ - && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then - md5sum -c << SHAR_EOF >/dev/null 2>&1 \ - || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed' -e87217fc3e71288102ba41fd81f71ec4 MultiSound.d/setdigital.c -SHAR_EOF - else - shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`" - test 2046 -eq "$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 <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <errno.h> -#include <unistd.h> -#include <asm/io.h> -#include <asm/types.h> -X -#define IREG_LOGDEVICE 0x07 -#define IREG_ACTIVATE 0x30 -#define LD_ACTIVATE 0x01 -#define LD_DISACTIVATE 0x00 -#define IREG_EECONTROL 0x3F -#define IREG_MEMBASEHI 0x40 -#define IREG_MEMBASELO 0x41 -#define IREG_MEMCONTROL 0x42 -#define IREG_MEMRANGEHI 0x43 -#define IREG_MEMRANGELO 0x44 -#define MEMTYPE_8BIT 0x00 -#define MEMTYPE_16BIT 0x02 -#define MEMTYPE_RANGE 0x00 -#define MEMTYPE_HIADDR 0x01 -#define IREG_IO0_BASEHI 0x60 -#define IREG_IO0_BASELO 0x61 -#define IREG_IO1_BASEHI 0x62 -#define IREG_IO1_BASELO 0x63 -#define IREG_IRQ_NUMBER 0x70 -#define IREG_IRQ_TYPE 0x71 -#define IRQTYPE_HIGH 0x02 -#define IRQTYPE_LOW 0x00 -#define IRQTYPE_LEVEL 0x01 -#define IRQTYPE_EDGE 0x00 -X -#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF)) -#define LOBYTE(w) ((BYTE)(w)) -#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8))) -X -typedef __u8 BYTE; -typedef __u16 USHORT; -typedef __u16 WORD; -X -static int config_port = -1; -X -static int msnd_write_cfg(int cfg, int reg, int value) -{ -X outb(reg, cfg); -X outb(value, cfg + 1); -X if (value != inb(cfg + 1)) { -X fprintf(stderr, "error: msnd_write_cfg: I/O error\n"); -X return -EIO; -X } -X return 0; -} -X -static int msnd_read_cfg(int cfg, int reg) -{ -X outb(reg, cfg); -X return inb(cfg + 1); -} -X -static int msnd_write_cfg_io0(int cfg, int num, WORD io) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) -X return -EIO; -X return 0; -} -X -static int msnd_read_cfg_io0(int cfg, int num, WORD *io) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X -X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO), -X msnd_read_cfg(cfg, IREG_IO0_BASEHI)); -X -X return 0; -} -X -static int msnd_write_cfg_io1(int cfg, int num, WORD io) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) -X return -EIO; -X return 0; -} -X -static int msnd_read_cfg_io1(int cfg, int num, WORD *io) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X -X *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO), -X msnd_read_cfg(cfg, IREG_IO1_BASEHI)); -X -X return 0; -} -X -static int msnd_write_cfg_irq(int cfg, int num, WORD irq) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) -X return -EIO; -X return 0; -} -X -static int msnd_read_cfg_irq(int cfg, int num, WORD *irq) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X -X *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER); -X -X return 0; -} -X -static int msnd_write_cfg_mem(int cfg, int num, int mem) -{ -X WORD wmem; -X -X mem >>= 8; -X mem &= 0xfff; -X wmem = (WORD)mem; -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) -X return -EIO; -X if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) -X return -EIO; -X return 0; -} -X -static int msnd_read_cfg_mem(int cfg, int num, int *mem) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X -X *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO), -X msnd_read_cfg(cfg, IREG_MEMBASEHI)); -X *mem <<= 8; -X -X return 0; -} -X -static int msnd_activate_logical(int cfg, int num) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) -X return -EIO; -X return 0; -} -X -static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_write_cfg_io0(cfg, num, io0)) -X return -EIO; -X if (msnd_write_cfg_io1(cfg, num, io1)) -X return -EIO; -X if (msnd_write_cfg_irq(cfg, num, irq)) -X return -EIO; -X if (msnd_write_cfg_mem(cfg, num, mem)) -X return -EIO; -X if (msnd_activate_logical(cfg, num)) -X return -EIO; -X return 0; -} -X -static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem) -{ -X if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) -X return -EIO; -X if (msnd_read_cfg_io0(cfg, num, io0)) -X return -EIO; -X if (msnd_read_cfg_io1(cfg, num, io1)) -X return -EIO; -X if (msnd_read_cfg_irq(cfg, num, irq)) -X return -EIO; -X if (msnd_read_cfg_mem(cfg, num, mem)) -X return -EIO; -X return 0; -} -X -static void usage(void) -{ -X fprintf(stderr, -X "\n" -X "pinnaclecfg 1.0\n" -X "\n" -X "usage: pinnaclecfg <config port> [device config]\n" -X "\n" -X "This is for use with the card in NON-PnP mode only.\n" -X "\n" -X "Available devices (not all available for Fiji):\n" -X "\n" -X " Device Description\n" -X " -------------------------------------------------------------------\n" -X " reset Reset all devices (i.e. disable)\n" -X " show Display current device configurations\n" -X "\n" -X " dsp <io> <irq> <mem> Audio device\n" -X " mpu <io> <irq> Internal Kurzweil synth\n" -X " ide <io0> <io1> <irq> On-board IDE controller\n" -X " joystick <io> Joystick port\n" -X "\n"); -X exit(1); -} -X -static int cfg_reset(void) -{ -X int i; -X -X for (i = 0; i < 4; ++i) -X msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0); -X -X return 0; -} -X -static int cfg_show(void) -{ -X int i; -X int count = 0; -X -X for (i = 0; i < 4; ++i) { -X WORD io0, io1, irq; -X int mem; -X msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem); -X switch (i) { -X case 0: -X if (io0 || irq || mem) { -X printf("dsp 0x%x %d 0x%x\n", io0, irq, mem); -X ++count; -X } -X break; -X case 1: -X if (io0 || irq) { -X printf("mpu 0x%x %d\n", io0, irq); -X ++count; -X } -X break; -X case 2: -X if (io0 || io1 || irq) { -X printf("ide 0x%x 0x%x %d\n", io0, io1, irq); -X ++count; -X } -X break; -X case 3: -X if (io0) { -X printf("joystick 0x%x\n", io0); -X ++count; -X } -X break; -X } -X } -X -X if (count == 0) -X fprintf(stderr, "no devices configured\n"); -X -X return 0; -} -X -static int cfg_dsp(int argc, char *argv[]) -{ -X int io, irq, mem; -X -X if (argc < 3 || -X sscanf(argv[0], "0x%x", &io) != 1 || -X sscanf(argv[1], "%d", &irq) != 1 || -X sscanf(argv[2], "0x%x", &mem) != 1) -X usage(); -X -X if (!(io == 0x290 || -X io == 0x260 || -X io == 0x250 || -X io == 0x240 || -X io == 0x230 || -X io == 0x220 || -X io == 0x210 || -X io == 0x3e0)) { -X fprintf(stderr, "error: io must be one of " -X "210, 220, 230, 240, 250, 260, 290, or 3E0\n"); -X usage(); -X } -X -X if (!(irq == 5 || -X irq == 7 || -X irq == 9 || -X irq == 10 || -X irq == 11 || -X irq == 12)) { -X fprintf(stderr, "error: irq must be one of " -X "5, 7, 9, 10, 11 or 12\n"); -X usage(); -X } -X -X if (!(mem == 0xb0000 || -X mem == 0xc8000 || -X mem == 0xd0000 || -X mem == 0xd8000 || -X mem == 0xe0000 || -X mem == 0xe8000)) { -X fprintf(stderr, "error: mem must be one of " -X "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); -X usage(); -X } -X -X return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem); -} -X -static int cfg_mpu(int argc, char *argv[]) -{ -X int io, irq; -X -X if (argc < 2 || -X sscanf(argv[0], "0x%x", &io) != 1 || -X sscanf(argv[1], "%d", &irq) != 1) -X usage(); -X -X return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0); -} -X -static int cfg_ide(int argc, char *argv[]) -{ -X int io0, io1, irq; -X -X if (argc < 3 || -X sscanf(argv[0], "0x%x", &io0) != 1 || -X sscanf(argv[0], "0x%x", &io1) != 1 || -X sscanf(argv[1], "%d", &irq) != 1) -X usage(); -X -X return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0); -} -X -static int cfg_joystick(int argc, char *argv[]) -{ -X int io; -X -X if (argc < 1 || -X sscanf(argv[0], "0x%x", &io) != 1) -X usage(); -X -X return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0); -} -X -int main(int argc, char *argv[]) -{ -X char *device; -X int rv = 0; -X -X --argc; ++argv; -X -X if (argc < 2) -X usage(); -X -X sscanf(argv[0], "0x%x", &config_port); -X if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) { -X fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n"); -X exit(1); -X } -X if (ioperm(config_port, 2, 1)) { -X perror("ioperm"); -X fprintf(stderr, "note: pinnaclecfg must be run as root\n"); -X exit(1); -X } -X device = argv[1]; -X -X argc -= 2; argv += 2; -X -X if (strcmp(device, "reset") == 0) -X rv = cfg_reset(); -X else if (strcmp(device, "show") == 0) -X rv = cfg_show(); -X else if (strcmp(device, "dsp") == 0) -X rv = cfg_dsp(argc, argv); -X else if (strcmp(device, "mpu") == 0) -X rv = cfg_mpu(argc, argv); -X else if (strcmp(device, "ide") == 0) -X rv = cfg_ide(argc, argv); -X else if (strcmp(device, "joystick") == 0) -X rv = cfg_joystick(argc, argv); -X else { -X fprintf(stderr, "error: unknown device %s\n", device); -X usage(); -X } -X -X if (rv) -X fprintf(stderr, "error: device configuration failed\n"); -X -X return 0; -} -SHAR_EOF - $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' && - chmod 0664 'MultiSound.d/pinnaclecfg.c' || - $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed' - if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ - && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then - md5sum -c << SHAR_EOF >/dev/null 2>&1 \ - || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed' -366bdf27f0db767a3c7921d0a6db20fe MultiSound.d/pinnaclecfg.c -SHAR_EOF - else - shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`" - test 10235 -eq "$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 <stdio.h> -#include <unistd.h> -#include <fcntl.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <sys/ioctl.h> -#include <sys/soundcard.h> -X -int main(int argc, char *argv[]) -{ -X int fd; -X -X if (argc != 2) { -X fprintf(stderr, "usage: msndreset <mixer device>\n"); -X exit(1); -X } -X -X if ((fd = open(argv[1], O_RDWR)) < 0) { -X perror(argv[1]); -X exit(1); -X } -X -X if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) { -X fprintf(stderr, "error: msnd ioctl reset failed\n"); -X perror("ioctl"); -X close(fd); -X exit(1); -X } -X -X close(fd); -X -X return 0; -} -SHAR_EOF - $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' && - chmod 0664 'MultiSound.d/msndreset.c' || - $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed' - if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \ - && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then - md5sum -c << SHAR_EOF >/dev/null 2>&1 \ - || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed' -c52f876521084e8eb25e12e01dcccb8a MultiSound.d/msndreset.c -SHAR_EOF - else - shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`" - test 1472 -eq "$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 deleted file mode 100644 index 2468ff82768..00000000000 --- a/Documentation/sound/oss/OPL3 +++ /dev/null @@ -1,6 +0,0 @@ -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 deleted file mode 100644 index 4cd5d9ab358..00000000000 --- a/Documentation/sound/oss/Opti +++ /dev/null @@ -1,218 +0,0 @@ -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.d/*.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 -------------------------------------------- -Config files in '/etc/modprobe.d/' are used as below: - -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 <stdio.h> -#include <fcntl.h> -#include <linux/soundcard.h> - -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 deleted file mode 100644 index 5c27229eec8..00000000000 --- a/Documentation/sound/oss/PAS16 +++ /dev/null @@ -1,162 +0,0 @@ -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 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=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2> - 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=<io>" 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.d/*.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.d/*.conf for pas2, sb and opl3 -respectively. diff --git a/Documentation/sound/oss/PSS b/Documentation/sound/oss/PSS deleted file mode 100644 index 187b9525e1f..00000000000 --- a/Documentation/sound/oss/PSS +++ /dev/null @@ -1,41 +0,0 @@ -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 deleted file mode 100644 index c84dd7597e6..00000000000 --- a/Documentation/sound/oss/PSS-updates +++ /dev/null @@ -1,88 +0,0 @@ - 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 deleted file mode 100644 index 4be259428a1..00000000000 --- a/Documentation/sound/oss/README.OSS +++ /dev/null @@ -1,1455 +0,0 @@ -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 <Ensoniq Soundscape (old)>, 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 deleted file mode 100644 index cdc039421a4..00000000000 --- a/Documentation/sound/oss/README.modules +++ /dev/null @@ -1,106 +0,0 @@ -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: - -<M> Sound card support - -<M> 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support - - I have SoundBlaster. Select your card from the list. - -<M> Generic OPL2/OPL3 FM synthesizer support -<M> 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.d/oss.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 -softdep sb post: 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/<kernel-version>/misc. Mine looks like: - -adlib_card.o # This is the generic OPLx driver -opl3.o # The OPL3 driver -sb.o # <<The SoundBlaster driver. Yours may differ.>> -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.d/*.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.d/*.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.d/*.conf files 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 deleted file mode 100644 index b6b77906b58..00000000000 --- a/Documentation/sound/oss/README.ymfsb +++ /dev/null @@ -1,107 +0,0 @@ -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 <jgarzik@pobox.com>'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 <breeze.nagano@nifty.ne.jp> - diff --git a/Documentation/sound/oss/SoundPro b/Documentation/sound/oss/SoundPro deleted file mode 100644 index 9d4db1f29d3..00000000000 --- a/Documentation/sound/oss/SoundPro +++ /dev/null @@ -1,105 +0,0 @@ -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 deleted file mode 100644 index b288d464ba8..00000000000 --- a/Documentation/sound/oss/Soundblaster +++ /dev/null @@ -1,53 +0,0 @@ -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+ deleted file mode 100644 index b93a6b734fc..00000000000 --- a/Documentation/sound/oss/Tropez+ +++ /dev/null @@ -1,26 +0,0 @@ -From: Paul Barton-Davis <pbd@op.net> - -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 deleted file mode 100644 index 68a5a46beb8..00000000000 --- a/Documentation/sound/oss/VIBRA16 +++ /dev/null @@ -1,80 +0,0 @@ -Sound Blaster 16X Vibra addendum --------------------------------- -by Marius Ilioaea <mariusi@protv.ro> - Stefan Laudat <stefan@asit.ro> - -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: - -<snip 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) - -<end of snipping> - - 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 deleted file mode 100644 index f4f3407cd81..00000000000 --- a/Documentation/sound/oss/WaveArtist +++ /dev/null @@ -1,170 +0,0 @@ - - (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 deleted file mode 100644 index 1a693e69d44..00000000000 --- a/Documentation/sound/oss/btaudio +++ /dev/null @@ -1,92 +0,0 @@ - -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 <kraxel@bytesex.org> diff --git a/Documentation/sound/oss/mwave b/Documentation/sound/oss/mwave deleted file mode 100644 index 5fbcb160927..00000000000 --- a/Documentation/sound/oss/mwave +++ /dev/null @@ -1,185 +0,0 @@ - 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 deleted file mode 100644 index 3ab391e7c29..00000000000 --- a/Documentation/sound/oss/oss-parameters.txt +++ /dev/null @@ -1,51 +0,0 @@ - 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: <io>,<irq>,<dma>,<dma2>,<type> - - aedsp16= [HW,OSS] Audio Excel DSP 16 - Format: <io>,<irq>,<dma>,<mss_io>,<mpu_io>,<mpu_irq> - See also header of sound/oss/aedsp16.c. - - dmasound= [HW,OSS] Sound subsystem buffers - - mpu401= [HW,OSS] - Format: <io>,<irq> - - opl3= [HW,OSS] - Format: <io> - - pas2= [HW,OSS] Format: - <io>,<irq>,<dma>,<dma16>,<sb_io>,<sb_irq>,<sb_dma>,<sb_dma16> - - pss= [HW,OSS] Personal Sound System (ECHO ESC614) - Format: - <io>,<mss_io>,<mss_irq>,<mss_dma>,<mpu_io>,<mpu_irq> - - sscape= [HW,OSS] - Format: <io>,<irq>,<dma>,<mpu_io>,<mpu_irq> - - trix= [HW,OSS] MediaTrix AudioTrix Pro - Format: - <io>,<irq>,<dma>,<dma2>,<sb_io>,<sb_irq>,<sb_dma>,<mpu_io>,<mpu_irq> - - uart401= [HW,OSS] - Format: <io>,<irq> - - uart6850= [HW,OSS] - Format: <io>,<irq> - - waveartist= [HW,OSS] - Format: <io>,<irq>,<dma>,<dma2> diff --git a/Documentation/sound/oss/ultrasound b/Documentation/sound/oss/ultrasound deleted file mode 100644 index eed331c738a..00000000000 --- a/Documentation/sound/oss/ultrasound +++ /dev/null @@ -1,30 +0,0 @@ -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 deleted file mode 100644 index 4c6cbdb3c54..00000000000 --- a/Documentation/sound/oss/vwsnd +++ /dev/null @@ -1,293 +0,0 @@ -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, <sys/soundcard.h> 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.h> - - ... - - 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); - } - |