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authorAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-08-05 17:04:01 -0300
committerAndré Fabian Silva Delgado <emulatorman@parabola.nu>2015-08-05 17:04:01 -0300
commit57f0f512b273f60d52568b8c6b77e17f5636edc0 (patch)
tree5e910f0e82173f4ef4f51111366a3f1299037a7b /Documentation/sound/alsa
Initial import
Diffstat (limited to 'Documentation/sound/alsa')
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt2312
-rw-r--r--Documentation/sound/alsa/Audigy-mixer.txt345
-rw-r--r--Documentation/sound/alsa/Audiophile-Usb.txt442
-rw-r--r--Documentation/sound/alsa/Bt87x.txt78
-rw-r--r--Documentation/sound/alsa/CMIPCI.txt254
-rw-r--r--Documentation/sound/alsa/Channel-Mapping-API.txt153
-rw-r--r--Documentation/sound/alsa/ControlNames.txt107
-rw-r--r--Documentation/sound/alsa/HD-Audio-Controls.txt116
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt314
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt863
-rw-r--r--Documentation/sound/alsa/Joystick.txt86
-rw-r--r--Documentation/sound/alsa/MIXART.txt100
-rw-r--r--Documentation/sound/alsa/OSS-Emulation.txt305
-rw-r--r--Documentation/sound/alsa/Procfile.txt234
-rw-r--r--Documentation/sound/alsa/README.maya44163
-rw-r--r--Documentation/sound/alsa/SB-Live-mixer.txt356
-rw-r--r--Documentation/sound/alsa/VIA82xx-mixer.txt8
-rw-r--r--Documentation/sound/alsa/alsa-parameters.txt135
-rw-r--r--Documentation/sound/alsa/compress_offload.txt234
-rw-r--r--Documentation/sound/alsa/emu10k1-jack.txt74
-rw-r--r--Documentation/sound/alsa/hda_codec.txt322
-rw-r--r--Documentation/sound/alsa/hdspm.txt362
-rw-r--r--Documentation/sound/alsa/powersave.txt41
-rw-r--r--Documentation/sound/alsa/seq_oss.html409
-rw-r--r--Documentation/sound/alsa/serial-u16550.txt88
-rw-r--r--Documentation/sound/alsa/soc/DAI.txt56
-rw-r--r--Documentation/sound/alsa/soc/DPCM.txt380
-rw-r--r--Documentation/sound/alsa/soc/clocking.txt51
-rw-r--r--Documentation/sound/alsa/soc/codec.txt179
-rw-r--r--Documentation/sound/alsa/soc/dapm.txt305
-rw-r--r--Documentation/sound/alsa/soc/jack.txt71
-rw-r--r--Documentation/sound/alsa/soc/machine.txt93
-rw-r--r--Documentation/sound/alsa/soc/overview.txt95
-rw-r--r--Documentation/sound/alsa/soc/platform.txt79
-rw-r--r--Documentation/sound/alsa/soc/pops_clicks.txt52
-rw-r--r--Documentation/sound/alsa/timestamping.txt200
36 files changed, 9462 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
new file mode 100644
index 000000000..5a8583abe
--- /dev/null
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -0,0 +1,2312 @@
+
+ Advanced Linux Sound Architecture - Driver
+ ==========================================
+ Configuration guide
+
+
+Kernel Configuration
+====================
+
+To enable ALSA support you need at least to build the kernel with
+primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS,
+you don't have to choose any of the OSS modules.
+
+Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and
+PCM supports if you want to run OSS applications with ALSA.
+
+If you want to support the WaveTable functionality on cards such as
+SB Live! then you need to enable "Sequencer support"
+(CONFIG_SND_SEQUENCER).
+
+To make ALSA debug messages more verbose, enable the "Verbose printk"
+and "Debug" options. To check for memory leaks, turn on "Debug memory"
+too. "Debug detection" will add checks for the detection of cards.
+
+Please note that all the ALSA ISA drivers support the Linux isapnp API
+(if the card supports ISA PnP). You don't need to configure the cards
+using isapnptools.
+
+
+Creating ALSA devices
+=====================
+
+This depends on your distribution, but normally you use the /dev/MAKEDEV
+script to create the necessary device nodes. On some systems you use a
+script named 'snddevices'.
+
+
+Module parameters
+=================
+
+The user can load modules with options. If the module supports more than
+one card and you have more than one card of the same type then you can
+specify multiple values for the option separated by commas.
+
+Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
+
+ Module snd
+ ----------
+
+ The core ALSA module. It is used by all ALSA card drivers.
+ It takes the following options which have global effects.
+
+ major - major number for sound driver
+ - Default: 116
+ cards_limit
+ - limiting card index for auto-loading (1-8)
+ - Default: 1
+ - For auto-loading more than one card, specify this
+ option together with snd-card-X aliases.
+ slots - Reserve the slot index for the given driver.
+ This option takes multiple strings.
+ See "Module Autoloading Support" section for details.
+ debug - Specifies the debug message level
+ (0 = disable debug prints, 1 = normal debug messages,
+ 2 = verbose debug messages)
+ This option appears only when CONFIG_SND_DEBUG=y.
+ This option can be dynamically changed via sysfs
+ /sys/modules/snd/parameters/debug file.
+
+ Module snd-pcm-oss
+ ------------------
+
+ The PCM OSS emulation module.
+ This module takes options which change the mapping of devices.
+
+ dsp_map - PCM device number maps assigned to the 1st OSS device.
+ - Default: 0
+ adsp_map - PCM device number maps assigned to the 2st OSS device.
+ - Default: 1
+ nonblock_open
+ - Don't block opening busy PCM devices. Default: 1
+
+ For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of
+ the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped
+ to PCM #0 of the card #0.
+ For changing the second or later card, specify the option with
+ commas, such like "dsp_map=0,1".
+
+ nonblock_open option is used to change the behavior of the PCM
+ regarding opening the device. When this option is non-zero,
+ opening a busy OSS PCM device won't be blocked but return
+ immediately with EAGAIN (just like O_NONBLOCK flag).
+
+ Module snd-rawmidi
+ ------------------
+
+ This module takes options which change the mapping of devices.
+ similar to those of the snd-pcm-oss module.
+
+ midi_map - MIDI device number maps assigned to the 1st OSS device.
+ - Default: 0
+ amidi_map - MIDI device number maps assigned to the 2st OSS device.
+ - Default: 1
+
+ Common parameters for top sound card modules
+ --------------------------------------------
+
+ Each of top level sound card module takes the following options.
+
+ index - index (slot #) of sound card
+ - Values: 0 through 31 or negative
+ - If nonnegative, assign that index number
+ - if negative, interpret as a bitmask of permissible
+ indices; the first free permitted index is assigned
+ - Default: -1
+ id - card ID (identifier or name)
+ - Can be up to 15 characters long
+ - Default: the card type
+ - A directory by this name is created under /proc/asound/
+ containing information about the card
+ - This ID can be used instead of the index number in
+ identifying the card
+ enable - enable card
+ - Default: enabled, for PCI and ISA PnP cards
+
+ Module snd-adlib
+ ----------------
+
+ Module for AdLib FM cards.
+
+ port - port # for OPL chip
+
+ This module supports multiple cards. It does not support autoprobe, so
+ the port must be specified. For actual AdLib FM cards it will be 0x388.
+ Note that this card does not have PCM support and no mixer; only FM
+ synthesis.
+
+ Make sure you have "sbiload" from the alsa-tools package available and,
+ after loading the module, find out the assigned ALSA sequencer port
+ number through "sbiload -l". Example output:
+
+ Port Client name Port name
+ 64:0 OPL2 FM synth OPL2 FM Port
+
+ Load the std.sb and drums.sb patches also supplied by sbiload:
+
+ sbiload -p 64:0 std.sb drums.sb
+
+ If you use this driver to drive an OPL3, you can use std.o3 and drums.o3
+ instead. To have the card produce sound, use aplaymidi from alsa-utils:
+
+ aplaymidi -p 64:0 foo.mid
+
+ Module snd-ad1816a
+ ------------------
+
+ Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips.
+
+ clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz)
+
+ This module supports multiple cards, autoprobe and PnP.
+
+ Module snd-ad1848
+ -----------------
+
+ Module for sound cards based on AD1848/AD1847/CS4248 ISA chips.
+
+ port - port # for AD1848 chip
+ irq - IRQ # for AD1848 chip
+ dma1 - DMA # for AD1848 chip (0,1,3)
+
+ This module supports multiple cards. It does not support autoprobe
+ thus main port must be specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ Module snd-ad1889
+ -----------------
+
+ Module for Analog Devices AD1889 chips.
+
+ ac97_quirk - AC'97 workaround for strange hardware
+ See the description of intel8x0 module for details.
+
+ This module supports multiple cards.
+
+ Module snd-ali5451
+ ------------------
+
+ Module for ALi M5451 PCI chip.
+
+ pcm_channels - Number of hardware channels assigned for PCM
+ spdif - Support SPDIF I/O
+ - Default: disabled
+
+ This module supports one chip and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-als100
+ -----------------
+
+ Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips.
+
+ This module supports multiple cards, autoprobe and PnP.
+
+ The power-management is supported.
+
+ Module snd-als300
+ -----------------
+
+ Module for Avance Logic ALS300 and ALS300+
+
+ This module supports multiple cards.
+
+ The power-management is supported.
+
+ Module snd-als4000
+ ------------------
+
+ Module for sound cards based on Avance Logic ALS4000 PCI chip.
+
+ joystick_port - port # for legacy joystick support.
+ 0 = disabled (default), 1 = auto-detect
+
+ This module supports multiple cards, autoprobe and PnP.
+
+ The power-management is supported.
+
+ Module snd-asihpi
+ -----------------
+
+ Module for AudioScience ASI soundcards
+
+ enable_hpi_hwdep - enable HPI hwdep for AudioScience soundcard
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-atiixp
+ -----------------
+
+ Module for ATI IXP 150/200/250/400 AC97 controllers.
+
+ ac97_clock - AC'97 clock (default = 48000)
+ ac97_quirk - AC'97 workaround for strange hardware
+ See "AC97 Quirk Option" section below.
+ ac97_codec - Workaround to specify which AC'97 codec
+ instead of probing. If this works for you
+ file a bug with your `lspci -vn` output.
+ -2 -- Force probing.
+ -1 -- Default behavior.
+ 0-2 -- Use the specified codec.
+ spdif_aclink - S/PDIF transfer over AC-link (default = 1)
+
+ This module supports one card and autoprobe.
+
+ ATI IXP has two different methods to control SPDIF output. One is
+ over AC-link and another is over the "direct" SPDIF output. The
+ implementation depends on the motherboard, and you'll need to
+ choose the correct one via spdif_aclink module option.
+
+ The power-management is supported.
+
+ Module snd-atiixp-modem
+ -----------------------
+
+ Module for ATI IXP 150/200/250 AC97 modem controllers.
+
+ This module supports one card and autoprobe.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ The power-management is supported.
+
+ Module snd-au8810, snd-au8820, snd-au8830
+ -----------------------------------------
+
+ Module for Aureal Vortex, Vortex2 and Advantage device.
+
+ pcifix - Control PCI workarounds
+ 0 = Disable all workarounds
+ 1 = Force the PCI latency of the Aureal card to 0xff
+ 2 = Force the Extend PCI#2 Internal Master for Efficient
+ Handling of Dummy Requests on the VIA KT133 AGP Bridge
+ 3 = Force both settings
+ 255 = Autodetect what is required (default)
+
+ This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware
+ EQ, mpu401, gameport. A3D and wavetable support are still in development.
+ Development and reverse engineering work is being coordinated at
+ http://savannah.nongnu.org/projects/openvortex/
+ SPDIF output has a copy of the AC97 codec output, unless you use the
+ "spdif" pcm device, which allows raw data passthru.
+ The hardware EQ hardware and SPDIF is only present in the Vortex2 and
+ Advantage.
+
+ Note: Some ALSA mixer applications don't handle the SPDIF sample rate
+ control correctly. If you have problems regarding this, try
+ another ALSA compliant mixer (alsamixer works).
+
+ Module snd-azt1605
+ ------------------
+
+ Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605
+ chipset.
+
+ port - port # for BASE (0x220,0x240,0x260,0x280)
+ wss_port - port # for WSS (0x530,0x604,0xe80,0xf40)
+ irq - IRQ # for WSS (7,9,10,11)
+ dma1 - DMA # for WSS playback (0,1,3)
+ dma2 - DMA # for WSS capture (0,1), -1 = disabled (default)
+ mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default)
+ mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default)
+ fm_port - port # for OPL3 (0x388), -1 = disabled (default)
+
+ This module supports multiple cards. It does not support autoprobe: port,
+ wss_port, irq and dma1 have to be specified. The other values are
+ optional.
+
+ "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240)
+ or the value stored in the card's EEPROM for cards that have an EEPROM and
+ their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can
+ be chosen freely from the options enumerated above.
+
+ If dma2 is specified and different from dma1, the card will operate in
+ full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to
+ enable capture since only channels 0 and 1 are available for capture.
+
+ Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0
+ mpu_port=0x330 mpu_irq=9 fm_port=0x388".
+
+ Whatever IRQ and DMA channels you pick, be sure to reserve them for
+ legacy ISA in your BIOS.
+
+ Module snd-azt2316
+ ------------------
+
+ Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316
+ chipset.
+
+ port - port # for BASE (0x220,0x240,0x260,0x280)
+ wss_port - port # for WSS (0x530,0x604,0xe80,0xf40)
+ irq - IRQ # for WSS (7,9,10,11)
+ dma1 - DMA # for WSS playback (0,1,3)
+ dma2 - DMA # for WSS capture (0,1), -1 = disabled (default)
+ mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default)
+ mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default)
+ fm_port - port # for OPL3 (0x388), -1 = disabled (default)
+
+ This module supports multiple cards. It does not support autoprobe: port,
+ wss_port, irq and dma1 have to be specified. The other values are
+ optional.
+
+ "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240)
+ or the value stored in the card's EEPROM for cards that have an EEPROM and
+ their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can
+ be chosen freely from the options enumerated above.
+
+ If dma2 is specified and different from dma1, the card will operate in
+ full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to
+ enable capture since only channels 0 and 1 are available for capture.
+
+ Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0
+ mpu_port=0x330 mpu_irq=9 fm_port=0x388".
+
+ Whatever IRQ and DMA channels you pick, be sure to reserve them for
+ legacy ISA in your BIOS.
+
+ Module snd-aw2
+ --------------
+
+ Module for Audiowerk2 sound card
+
+ This module supports multiple cards.
+
+ Module snd-azt2320
+ ------------------
+
+ Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only).
+
+ This module supports multiple cards, PnP and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-azt3328
+ ------------------
+
+ Module for sound cards based on Aztech AZF3328 PCI chip.
+
+ joystick - Enable joystick (default off)
+
+ This module supports multiple cards.
+
+ Module snd-bt87x
+ ----------------
+
+ Module for video cards based on Bt87x chips.
+
+ digital_rate - Override the default digital rate (Hz)
+ load_all - Load the driver even if the card model isn't known
+
+ This module supports multiple cards.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ Module snd-ca0106
+ -----------------
+
+ Module for Creative Audigy LS and SB Live 24bit
+
+ This module supports multiple cards.
+
+
+ Module snd-cmi8330
+ ------------------
+
+ Module for sound cards based on C-Media CMI8330 ISA chips.
+
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ with isapnp=0, the following options are available:
+
+ wssport - port # for CMI8330 chip (WSS)
+ wssirq - IRQ # for CMI8330 chip (WSS)
+ wssdma - first DMA # for CMI8330 chip (WSS)
+ sbport - port # for CMI8330 chip (SB16)
+ sbirq - IRQ # for CMI8330 chip (SB16)
+ sbdma8 - 8bit DMA # for CMI8330 chip (SB16)
+ sbdma16 - 16bit DMA # for CMI8330 chip (SB16)
+ fmport - (optional) OPL3 I/O port
+ mpuport - (optional) MPU401 I/O port
+ mpuirq - (optional) MPU401 irq #
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-cmipci
+ -----------------
+
+ Module for C-Media CMI8338/8738/8768/8770 PCI sound cards.
+
+ mpu_port - port address of MIDI interface (8338 only):
+ 0x300,0x310,0x320,0x330 = legacy port,
+ 0 = disable (default)
+ fm_port - port address of OPL-3 FM synthesizer (8x38 only):
+ 0x388 = legacy port,
+ 1 = integrated PCI port (default on 8738),
+ 0 = disable
+ soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only)
+ (default = 1)
+ joystick_port - Joystick port address (0 = disable, 1 = auto-detect)
+
+ This module supports autoprobe and multiple cards.
+
+ The power-management is supported.
+
+ Module snd-cs4231
+ -----------------
+
+ Module for sound cards based on CS4231 ISA chips.
+
+ port - port # for CS4231 chip
+ mpu_port - port # for MPU-401 UART (optional), -1 = disable
+ irq - IRQ # for CS4231 chip
+ mpu_irq - IRQ # for MPU-401 UART
+ dma1 - first DMA # for CS4231 chip
+ dma2 - second DMA # for CS4231 chip
+
+ This module supports multiple cards. This module does not support autoprobe
+ thus main port must be specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ Module snd-cs4236
+ -----------------
+
+ Module for sound cards based on CS4232/CS4232A,
+ CS4235/CS4236/CS4236B/CS4237B/
+ CS4238B/CS4239 ISA chips.
+
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ with isapnp=0, the following options are available:
+
+ port - port # for CS4236 chip (PnP setup - 0x534)
+ cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00)
+ mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
+ fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable
+ irq - IRQ # for CS4236 chip (5,7,9,11,12,15)
+ mpu_irq - IRQ # for MPU-401 UART (9,11,12,15)
+ dma1 - first DMA # for CS4236 chip (0,1,3)
+ dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable
+
+ This module supports multiple cards. This module does not support autoprobe
+ (if ISA PnP is not used) thus main port and control port must be
+ specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ This module is aliased as snd-cs4232 since it provides the old
+ snd-cs4232 functionality, too.
+
+ Module snd-cs4281
+ -----------------
+
+ Module for Cirrus Logic CS4281 soundchip.
+
+ dual_codec - Secondary codec ID (0 = disable, default)
+
+ This module supports multiple cards.
+
+ The power-management is supported.
+
+ Module snd-cs46xx
+ -----------------
+
+ Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/
+ CS4624/CS4630/CS4280 PCI chips.
+
+ external_amp - Force to enable external amplifier.
+ thinkpad - Force to enable Thinkpad's CLKRUN control.
+ mmap_valid - Support OSS mmap mode (default = 0).
+
+ This module supports multiple cards and autoprobe.
+ Usually external amp and CLKRUN controls are detected automatically
+ from PCI sub vendor/device ids. If they don't work, give the options
+ above explicitly.
+
+ The power-management is supported.
+
+ Module snd-cs5530
+ _________________
+
+ Module for Cyrix/NatSemi Geode 5530 chip.
+
+ Module snd-cs5535audio
+ ----------------------
+
+ Module for multifunction CS5535 companion PCI device
+
+ The power-management is supported.
+
+ Module snd-ctxfi
+ ----------------
+
+ Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips)
+ * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series
+ * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
+ * Creative Sound Blaster X-Fi Titanium Professional Audio
+ * Creative Sound Blaster X-Fi Titanium
+ * Creative Sound Blaster X-Fi Elite Pro
+ * Creative Sound Blaster X-Fi Platinum
+ * Creative Sound Blaster X-Fi Fatal1ty
+ * Creative Sound Blaster X-Fi XtremeGamer
+ * Creative Sound Blaster X-Fi XtremeMusic
+
+ reference_rate - reference sample rate, 44100 or 48000 (default)
+ multiple - multiple to ref. sample rate, 1 or 2 (default)
+ subsystem - override the PCI SSID for probing; the value
+ consists of SSVID << 16 | SSDID. The default is
+ zero, which means no override.
+
+ This module supports multiple cards.
+
+ Module snd-darla20
+ ------------------
+
+ Module for Echoaudio Darla20
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-darla24
+ ------------------
+
+ Module for Echoaudio Darla24
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-dt019x
+ -----------------
+
+ Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP
+ only)
+
+ This module supports multiple cards. This module is enabled only with
+ ISA PnP support.
+
+ The power-management is supported.
+
+ Module snd-dummy
+ ----------------
+
+ Module for the dummy sound card. This "card" doesn't do any output
+ or input, but you may use this module for any application which
+ requires a sound card (like RealPlayer).
+
+ pcm_devs - Number of PCM devices assigned to each card
+ (default = 1, up to 4)
+ pcm_substreams - Number of PCM substreams assigned to each PCM
+ (default = 8, up to 128)
+ hrtimer - Use hrtimer (=1, default) or system timer (=0)
+ fake_buffer - Fake buffer allocations (default = 1)
+
+ When multiple PCM devices are created, snd-dummy gives different
+ behavior to each PCM device:
+ 0 = interleaved with mmap support
+ 1 = non-interleaved with mmap support
+ 2 = interleaved without mmap
+ 3 = non-interleaved without mmap
+
+ As default, snd-dummy drivers doesn't allocate the real buffers
+ but either ignores read/write or mmap a single dummy page to all
+ buffer pages, in order to save the resources. If your apps need
+ the read/ written buffer data to be consistent, pass fake_buffer=0
+ option.
+
+ The power-management is supported.
+
+ Module snd-echo3g
+ -----------------
+
+ Module for Echoaudio 3G cards (Gina3G/Layla3G)
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-emu10k1
+ ------------------
+
+ Module for EMU10K1/EMU10k2 based PCI sound cards.
+ * Sound Blaster Live!
+ * Sound Blaster PCI 512
+ * Emu APS (partially supported)
+ * Sound Blaster Audigy
+
+ extin - bitmap of available external inputs for FX8010 (see bellow)
+ extout - bitmap of available external outputs for FX8010 (see bellow)
+ seq_ports - allocated sequencer ports (4 by default)
+ max_synth_voices - limit of voices used for wavetable (64 by default)
+ max_buffer_size - specifies the maximum size of wavetable/pcm buffers
+ given in MB unit. Default value is 128.
+ enable_ir - enable IR
+
+ This module supports multiple cards and autoprobe.
+
+ Input & Output configurations [extin/extout]
+ * Creative Card wo/Digital out [0x0003/0x1f03]
+ * Creative Card w/Digital out [0x0003/0x1f0f]
+ * Creative Card w/Digital CD in [0x000f/0x1f0f]
+ * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3]
+ * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf]
+ * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf]
+ * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f]
+ * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f]
+ * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f]
+ * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff]
+ * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff]
+ * Creative Card all ins and outs [0x3fff/0x7fff]
+
+ The power-management is supported.
+
+ Module snd-emu10k1x
+ -------------------
+
+ Module for Creative Emu10k1X (SB Live Dell OEM version)
+
+ This module supports multiple cards.
+
+ Module snd-ens1370
+ ------------------
+
+ Module for Ensoniq AudioPCI ES1370 PCI sound cards.
+ * SoundBlaster PCI 64
+ * SoundBlaster PCI 128
+
+ joystick - Enable joystick (default off)
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-ens1371
+ ------------------
+
+ Module for Ensoniq AudioPCI ES1371 PCI sound cards.
+ * SoundBlaster PCI 64
+ * SoundBlaster PCI 128
+ * SoundBlaster Vibra PCI
+
+ joystick_port - port # for joystick (0x200,0x208,0x210,0x218),
+ 0 = disable (default), 1 = auto-detect
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-es1688
+ -----------------
+
+ Module for ESS AudioDrive ES-1688 and ES-688 sound cards.
+
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+ mpu_irq - IRQ # for MPU-401 port (5,7,9,10)
+ fm_port - port # for OPL3 (option; share the same port as default)
+
+ with isapnp=0, the following additional options are available:
+ port - port # for ES-1688 chip (0x220,0x240,0x260)
+ irq - IRQ # for ES-1688 chip (5,7,9,10)
+ dma8 - DMA # for ES-1688 chip (0,1,3)
+
+ This module supports multiple cards and autoprobe (without MPU-401 port)
+ and PnP with the ES968 chip.
+
+ Module snd-es18xx
+ -----------------
+
+ Module for ESS AudioDrive ES-18xx sound cards.
+
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ with isapnp=0, the following options are available:
+
+ port - port # for ES-18xx chip (0x220,0x240,0x260)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+ fm_port - port # for FM (optional, not used)
+ irq - IRQ # for ES-18xx chip (5,7,9,10)
+ dma1 - first DMA # for ES-18xx chip (0,1,3)
+ dma2 - first DMA # for ES-18xx chip (0,1,3)
+
+ This module supports multiple cards, ISA PnP and autoprobe (without MPU-401
+ port if native ISA PnP routines are not used).
+ When dma2 is equal with dma1, the driver works as half-duplex.
+
+ The power-management is supported.
+
+ Module snd-es1938
+ -----------------
+
+ Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips.
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-es1968
+ -----------------
+
+ Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips.
+
+ total_bufsize - total buffer size in kB (1-4096kB)
+ pcm_substreams_p - playback channels (1-8, default=2)
+ pcm_substreams_c - capture channels (1-8, default=0)
+ clock - clock (0 = auto-detection)
+ use_pm - support the power-management (0 = off, 1 = on,
+ 2 = auto (default))
+ enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default))
+ joystick - enable joystick (default off)
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-fm801
+ ----------------
+
+ Module for ForteMedia FM801 based PCI sound cards.
+
+ tea575x_tuner - Enable TEA575x tuner
+ - 1 = MediaForte 256-PCS
+ - 2 = MediaForte 256-PCPR
+ - 3 = MediaForte 64-PCR
+ - High 16-bits are video (radio) device number + 1
+ - example: 0x10002 (MediaForte 256-PCPR, device 1)
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-gina20
+ -----------------
+
+ Module for Echoaudio Gina20
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-gina24
+ -----------------
+
+ Module for Echoaudio Gina24
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-gusclassic
+ ---------------------
+
+ Module for Gravis UltraSound Classic sound card.
+
+ port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ This module supports multiple cards and autoprobe.
+
+ Module snd-gusextreme
+ ---------------------
+
+ Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card.
+
+ port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260)
+ gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable
+ irq - IRQ # for ES-1688 chip (5,7,9,10)
+ gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ mpu_irq - IRQ # for MPU-401 port (5,7,9,10)
+ dma8 - DMA # for ES-1688 chip (0,1,3)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ This module supports multiple cards and autoprobe (without MPU-401 port).
+
+ Module snd-gusmax
+ -----------------
+
+ Module for Gravis UltraSound MAX sound card.
+
+ port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ This module supports multiple cards and autoprobe.
+
+ Module snd-hda-intel
+ --------------------
+
+ Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10,
+ PCH, SCH),
+ ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620,
+ RV630, RV635, RV670, RV770,
+ VIA VT8251/VT8237A,
+ SIS966, ULI M5461
+
+ [Multiple options for each card instance]
+ model - force the model name
+ position_fix - Fix DMA pointer
+ -1 = system default: choose appropriate one per controller
+ hardware
+ 0 = auto: falls back to LPIB when POSBUF doesn't work
+ 1 = use LPIB
+ 2 = POSBUF: use position buffer
+ 3 = VIACOMBO: VIA-specific workaround for capture
+ 4 = COMBO: use LPIB for playback, auto for capture stream
+ probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
+ When the bit 8 (0x100) is set, the lower 8 bits are used
+ as the "fixed" codec slots; i.e. the driver probes the
+ slots regardless what hardware reports back
+ probe_only - Only probing and no codec initialization (default=off);
+ Useful to check the initial codec status for debugging
+ bdl_pos_adj - Specifies the DMA IRQ timing delay in samples.
+ Passing -1 will make the driver to choose the appropriate
+ value based on the controller chip.
+ patch - Specifies the early "patch" files to modify the HD-audio
+ setup before initializing the codecs. This option is
+ available only when CONFIG_SND_HDA_PATCH_LOADER=y is set.
+ See HD-Audio.txt for details.
+ beep_mode - Selects the beep registration mode (0=off, 1=on); default
+ value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig.
+
+ [Single (global) options]
+ single_cmd - Use single immediate commands to communicate with
+ codecs (for debugging only)
+ enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
+ power_save - Automatic power-saving timeout (in second, 0 =
+ disable)
+ power_save_controller - Reset HD-audio controller in power-saving mode
+ (default = on)
+ align_buffer_size - Force rounding of buffer/period sizes to multiples
+ of 128 bytes. This is more efficient in terms of memory
+ access but isn't required by the HDA spec and prevents
+ users from specifying exact period/buffer sizes.
+ (default = on)
+ snoop - Enable/disable snooping (default = on)
+
+ This module supports multiple cards and autoprobe.
+
+ See Documentation/sound/alsa/HD-Audio.txt for more details about
+ HD-audio driver.
+
+ Each codec may have a model table for different configurations.
+ If your machine isn't listed there, the default (usually minimal)
+ configuration is set up. You can pass "model=<name>" option to
+ specify a certain model in such a case. There are different
+ models depending on the codec chip. The list of available models
+ is found in HD-Audio-Models.txt
+
+ The model name "generic" is treated as a special case. When this
+ model is given, the driver uses the generic codec parser without
+ "codec-patch". It's sometimes good for testing and debugging.
+
+ If the default configuration doesn't work and one of the above
+ matches with your device, report it together with alsa-info.sh
+ output (with --no-upload option) to kernel bugzilla or alsa-devel
+ ML (see the section "Links and Addresses").
+
+ power_save and power_save_controller options are for power-saving
+ mode. See powersave.txt for details.
+
+ Note 2: If you get click noises on output, try the module option
+ position_fix=1 or 2. position_fix=1 will use the SD_LPIB
+ register value without FIFO size correction as the current
+ DMA pointer. position_fix=2 will make the driver to use
+ the position buffer instead of reading SD_LPIB register.
+ (Usually SD_LPIB register is more accurate than the
+ position buffer.)
+
+ position_fix=3 is specific to VIA devices. The position
+ of the capture stream is checked from both LPIB and POSBUF
+ values. position_fix=4 is a combination mode, using LPIB
+ for playback and POSBUF for capture.
+
+ NB: If you get many "azx_get_response timeout" messages at
+ loading, it's likely a problem of interrupts (e.g. ACPI irq
+ routing). Try to boot with options like "pci=noacpi". Also, you
+ can try "single_cmd=1" module option. This will switch the
+ communication method between HDA controller and codecs to the
+ single immediate commands instead of CORB/RIRB. Basically, the
+ single command mode is provided only for BIOS, and you won't get
+ unsolicited events, too. But, at least, this works independently
+ from the irq. Remember this is a last resort, and should be
+ avoided as much as possible...
+
+ MORE NOTES ON "azx_get_response timeout" PROBLEMS:
+ On some hardware, you may need to add a proper probe_mask option
+ to avoid the "azx_get_response timeout" problem above, instead.
+ This occurs when the access to non-existing or non-working codec slot
+ (likely a modem one) causes a stall of the communication via HD-audio
+ bus. You can see which codec slots are probed by enabling
+ CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec
+ proc files. Then limit the slots to probe by probe_mask option.
+ For example, probe_mask=1 means to probe only the first slot, and
+ probe_mask=4 means only the third slot.
+
+ The power-management is supported.
+
+ Module snd-hdsp
+ ---------------
+
+ Module for RME Hammerfall DSP audio interface(s)
+
+ This module supports multiple cards.
+
+ Note: The firmware data can be automatically loaded via hotplug
+ when CONFIG_FW_LOADER is set. Otherwise, you need to load
+ the firmware via hdsploader utility included in alsa-tools
+ package.
+ The firmware data is found in alsa-firmware package.
+
+ Note: snd-page-alloc module does the job which snd-hammerfall-mem
+ module did formerly. It will allocate the buffers in advance
+ when any HDSP cards are found. To make the buffer
+ allocation sure, load snd-page-alloc module in the early
+ stage of boot sequence. See "Early Buffer Allocation"
+ section.
+
+ Module snd-hdspm
+ ----------------
+
+ Module for RME HDSP MADI board.
+
+ precise_ptr - Enable precise pointer, or disable.
+ line_outs_monitor - Send playback streams to analog outs by default.
+ enable_monitor - Enable Analog Out on Channel 63/64 by default.
+
+ See hdspm.txt for details.
+
+ Module snd-ice1712
+ ------------------
+
+ Module for Envy24 (ICE1712) based PCI sound cards.
+ * MidiMan M Audio Delta 1010
+ * MidiMan M Audio Delta 1010LT
+ * MidiMan M Audio Delta DiO 2496
+ * MidiMan M Audio Delta 66
+ * MidiMan M Audio Delta 44
+ * MidiMan M Audio Delta 410
+ * MidiMan M Audio Audiophile 2496
+ * TerraTec EWS 88MT
+ * TerraTec EWS 88D
+ * TerraTec EWX 24/96
+ * TerraTec DMX 6Fire
+ * TerraTec Phase 88
+ * Hoontech SoundTrack DSP 24
+ * Hoontech SoundTrack DSP 24 Value
+ * Hoontech SoundTrack DSP 24 Media 7.1
+ * Event Electronics, EZ8
+ * Digigram VX442
+ * Lionstracs, Mediastaton
+ * Terrasoniq TS 88
+
+ model - Use the given board model, one of the following:
+ delta1010, dio2496, delta66, delta44, audiophile, delta410,
+ delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d,
+ dmx6fire, dsp24, dsp24_value, dsp24_71, ez8,
+ phase88, mediastation
+ omni - Omni I/O support for MidiMan M-Audio Delta44/66
+ cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver)
+ in msec resolution, default value is 500 (0.5 sec)
+
+ This module supports multiple cards and autoprobe. Note: The consumer part
+ is not used with all Envy24 based cards (for example in the MidiMan Delta
+ serie).
+
+ Note: The supported board is detected by reading EEPROM or PCI
+ SSID (if EEPROM isn't available). You can override the
+ model by passing "model" module option in case that the
+ driver isn't configured properly or you want to try another
+ type for testing.
+
+ Module snd-ice1724
+ ------------------
+
+ Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+ * MidiMan M Audio Revolution 5.1
+ * MidiMan M Audio Revolution 7.1
+ * MidiMan M Audio Audiophile 192
+ * AMP Ltd AUDIO2000
+ * TerraTec Aureon 5.1 Sky
+ * TerraTec Aureon 7.1 Space
+ * TerraTec Aureon 7.1 Universe
+ * TerraTec Phase 22
+ * TerraTec Phase 28
+ * AudioTrak Prodigy 7.1
+ * AudioTrak Prodigy 7.1 LT
+ * AudioTrak Prodigy 7.1 XT
+ * AudioTrak Prodigy 7.1 HIFI
+ * AudioTrak Prodigy 7.1 HD2
+ * AudioTrak Prodigy 192
+ * Pontis MS300
+ * Albatron K8X800 Pro II
+ * Chaintech ZNF3-150
+ * Chaintech ZNF3-250
+ * Chaintech 9CJS
+ * Chaintech AV-710
+ * Shuttle SN25P
+ * Onkyo SE-90PCI
+ * Onkyo SE-200PCI
+ * ESI Juli@
+ * ESI Maya44
+ * Hercules Fortissimo IV
+ * EGO-SYS WaveTerminal 192M
+
+ model - Use the given board model, one of the following:
+ revo51, revo71, amp2000, prodigy71, prodigy71lt,
+ prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192,
+ juli, aureon51, aureon71, universe, ap192, k8x800,
+ phase22, phase28, ms300, av710, se200pci, se90pci,
+ fortissimo4, sn25p, WT192M, maya44
+
+ This module supports multiple cards and autoprobe.
+
+ Note: The supported board is detected by reading EEPROM or PCI
+ SSID (if EEPROM isn't available). You can override the
+ model by passing "model" module option in case that the
+ driver isn't configured properly or you want to try another
+ type for testing.
+
+ Module snd-indigo
+ -----------------
+
+ Module for Echoaudio Indigo
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-indigodj
+ -------------------
+
+ Module for Echoaudio Indigo DJ
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-indigoio
+ -------------------
+
+ Module for Echoaudio Indigo IO
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-intel8x0
+ -------------------
+
+ Module for AC'97 motherboards from Intel and compatibles.
+ * Intel i810/810E, i815, i820, i830, i84x, MX440
+ ICH5, ICH6, ICH7, 6300ESB, ESB2
+ * SiS 7012 (SiS 735)
+ * NVidia NForce, NForce2, NForce3, MCP04, CK804
+ CK8, CK8S, MCP501
+ * AMD AMD768, AMD8111
+ * ALi m5455
+
+ ac97_clock - AC'97 codec clock base (0 = auto-detect)
+ ac97_quirk - AC'97 workaround for strange hardware
+ See "AC97 Quirk Option" section below.
+ buggy_irq - Enable workaround for buggy interrupts on some
+ motherboards (default yes on nForce chips,
+ otherwise off)
+ buggy_semaphore - Enable workaround for hardware with buggy
+ semaphores (e.g. on some ASUS laptops)
+ (default off)
+ spdif_aclink - Use S/PDIF over AC-link instead of direct connection
+ from the controller chip
+ (0 = off, 1 = on, -1 = default)
+
+ This module supports one chip and autoprobe.
+
+ Note: the latest driver supports auto-detection of chip clock.
+ if you still encounter too fast playback, specify the clock
+ explicitly via the module option "ac97_clock=41194".
+
+ Joystick/MIDI ports are not supported by this driver. If your
+ motherboard has these devices, use the ns558 or snd-mpu401
+ modules, respectively.
+
+ The power-management is supported.
+
+ Module snd-intel8x0m
+ --------------------
+
+ Module for Intel ICH (i8x0) chipset MC97 modems.
+ * Intel i810/810E, i815, i820, i830, i84x, MX440
+ ICH5, ICH6, ICH7
+ * SiS 7013 (SiS 735)
+ * NVidia NForce, NForce2, NForce2s, NForce3
+ * AMD AMD8111
+ * ALi m5455
+
+ ac97_clock - AC'97 codec clock base (0 = auto-detect)
+
+ This module supports one card and autoprobe.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ The power-management is supported.
+
+ Module snd-interwave
+ --------------------
+
+ Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32
+ and other sound cards based on AMD InterWave (tm) chip.
+
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+ pcm_voices - reserved PCM voices for the synthesizer (default 2)
+ effect - 1 = InterWave effects enable (default 0);
+ requires 8 voices
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ with isapnp=0, the following options are available:
+
+ port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for InterWave chip (3,5,9,11,12,15)
+ dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
+ dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+
+ This module supports multiple cards, autoprobe and ISA PnP.
+
+ Module snd-interwave-stb
+ ------------------------
+
+ Module for UltraSound 32-Pro (sound card from STB used by Compaq)
+ and other sound cards based on AMD InterWave (tm) chip with TEA6330T
+ circuit for extended control of bass, treble and master volume.
+
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+ pcm_voices - reserved PCM voices for the synthesizer (default 2)
+ effect - 1 = InterWave effects enable (default 0);
+ requires 8 voices
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ with isapnp=0, the following options are available:
+
+ port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+ port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380)
+ irq - IRQ # for InterWave chip (3,5,9,11,12,15)
+ dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
+ dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+
+ This module supports multiple cards, autoprobe and ISA PnP.
+
+ Module snd-jazz16
+ -------------------
+
+ Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips:
+ MVD1216 + MVA416 + MVA514.
+
+ port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for SB DSP chip (3,5,7,9,10,15)
+ dma8 - DMA # for SB DSP chip (1,3)
+ dma16 - DMA # for SB DSP chip (5,7)
+ mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330)
+ mpu_irq - MPU-401 irq # (2,3,5,7)
+
+ This module supports multiple cards.
+
+ Module snd-korg1212
+ -------------------
+
+ Module for Korg 1212 IO PCI card
+
+ This module supports multiple cards.
+
+ Module snd-layla20
+ ------------------
+
+ Module for Echoaudio Layla20
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-layla24
+ ------------------
+
+ Module for Echoaudio Layla24
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-lola
+ ---------------
+
+ Module for Digigram Lola PCI-e boards
+
+ This module supports multiple cards.
+
+ Module snd-lx6464es
+ -------------------
+
+ Module for Digigram LX6464ES boards
+
+ This module supports multiple cards.
+
+ Module snd-maestro3
+ -------------------
+
+ Module for Allegro/Maestro3 chips
+
+ external_amp - enable external amp (enabled by default)
+ amp_gpio - GPIO pin number for external amp (0-15) or
+ -1 for default pin (8 for allegro, 1 for
+ others)
+
+ This module supports autoprobe and multiple chips.
+
+ Note: the binding of amplifier is dependent on hardware.
+ If there is no sound even though all channels are unmuted, try to
+ specify other gpio connection via amp_gpio option.
+ For example, a Panasonic notebook might need "amp_gpio=0x0d"
+ option.
+
+ The power-management is supported.
+
+ Module snd-mia
+ ---------------
+
+ Module for Echoaudio Mia
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-miro
+ ---------------
+
+ Module for Miro soundcards: miroSOUND PCM 1 pro,
+ miroSOUND PCM 12,
+ miroSOUND PCM 20 Radio.
+
+ port - Port # (0x530,0x604,0xe80,0xf40)
+ irq - IRQ # (5,7,9,10,11)
+ dma1 - 1st dma # (0,1,3)
+ dma2 - 2nd dma # (0,1)
+ mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330)
+ mpu_irq - MPU-401 irq # (5,7,9,10)
+ fm_port - FM Port # (0x388)
+ wss - enable WSS mode
+ ide - enable onboard ide support
+
+ Module snd-mixart
+ -----------------
+
+ Module for Digigram miXart8 sound cards.
+
+ This module supports multiple cards.
+ Note: One miXart8 board will be represented as 4 alsa cards.
+ See MIXART.txt for details.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via mixartloader utility in alsa-tools package.
+
+ Module snd-mona
+ ---------------
+
+ Module for Echoaudio Mona
+
+ This module supports multiple cards.
+ The driver requires the firmware loader support on kernel.
+
+ Module snd-mpu401
+ -----------------
+
+ Module for MPU-401 UART devices.
+
+ port - port number or -1 (disable)
+ irq - IRQ number or -1 (disable)
+ pnp - PnP detection - 0 = disable, 1 = enable (default)
+
+ This module supports multiple devices and PnP.
+
+ Module snd-msnd-classic
+ -----------------------
+
+ Module for Turtle Beach MultiSound Classic, Tahiti or Monterey
+ soundcards.
+
+ io - Port # for msnd-classic card
+ irq - IRQ # for msnd-classic card
+ mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000,
+ 0xe0000 or 0xe8000)
+ write_ndelay - enable write ndelay (default = 1)
+ calibrate_signal - calibrate signal (default = 0)
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+ digital - Digital daughterboard present (default = 0)
+ cfg - Config port (0x250, 0x260 or 0x270) default = PnP
+ reset - Reset all devices
+ mpu_io - MPU401 I/O port
+ mpu_irq - MPU401 irq#
+ ide_io0 - IDE port #0
+ ide_io1 - IDE port #1
+ ide_irq - IDE irq#
+ joystick_io - Joystick I/O port
+
+ The driver requires firmware files "/*(DEBLOBBED)*/" and
+ "/*(DEBLOBBED)*/" 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 "/*(DEBLOBBED)*/" and
+ "/*(DEBLOBBED)*/" 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 "/*(DEBLOBBED)*/" 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)
+ /*(DEBLOBBED)*/
+ 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.
+
+ /*(DEBLOBBED)*/.
+
+ Module snd-sonicvibes
+ ---------------------
+
+ Module for S3 SonicVibes PCI sound cards.
+ * PINE Schubert 32 PCI
+
+ reverb - Reverb Enable - 1 = enable, 0 = disable (default)
+ - SoundCard must have onboard SRAM for this.
+ mge - Mic Gain Enable - 1 = enable, 0 = disable (default)
+
+ This module supports multiple cards and autoprobe.
+
+ Module snd-serial-u16550
+ ------------------------
+
+ Module for UART16550A serial MIDI ports.
+
+ port - port # for UART16550A chip
+ irq - IRQ # for UART16550A chip, -1 = poll mode
+ speed - speed in bauds (9600,19200,38400,57600,115200)
+ 38400 = default
+ base - base for divisor in bauds (57600,115200,230400,460800)
+ 115200 = default
+ outs - number of MIDI ports in a serial port (1-4)
+ 1 = default
+ adaptor - Type of adaptor.
+ 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A,
+ 3 = MS-124W M/B, 4 = Generic
+
+ This module supports multiple cards. This module does not support autoprobe
+ thus the main port must be specified!!! Other options are optional.
+
+ Module snd-trident
+ ------------------
+
+ Module for Trident 4DWave DX/NX sound cards.
+ * Best Union Miss Melody 4DWave PCI
+ * HIS 4DWave PCI
+ * Warpspeed ONSpeed 4DWave PCI
+ * AzTech PCI 64-Q3D
+ * Addonics SV 750
+ * CHIC True Sound 4Dwave
+ * Shark Predator4D-PCI
+ * Jaton SonicWave 4D
+ * SiS SI7018 PCI Audio
+ * Hoontech SoundTrack Digital 4DWave NX
+
+ pcm_channels - max channels (voices) reserved for PCM
+ wavetable_size - max wavetable size in kB (4-?kb)
+
+ This module supports multiple cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-ua101
+ ----------------
+
+ Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces.
+
+ This module supports multiple devices, autoprobe and hotplugging.
+
+ Module snd-usb-audio
+ --------------------
+
+ Module for USB audio and USB MIDI devices.
+
+ vid - Vendor ID for the device (optional)
+ pid - Product ID for the device (optional)
+ nrpacks - Max. number of packets per URB (default: 8)
+ device_setup - Device specific magic number (optional)
+ - Influence depends on the device
+ - Default: 0x0000
+ ignore_ctl_error - Ignore any USB-controller regarding mixer
+ interface (default: no)
+
+ This module supports multiple devices, autoprobe and hotplugging.
+
+ NB: nrpacks parameter can be modified dynamically via sysfs.
+ Don't put the value over 20. Changing via sysfs has no sanity
+ check.
+ NB: ignore_ctl_error=1 may help when you get an error at accessing
+ the mixer element such as URB error -22. This happens on some
+ buggy USB device or the controller.
+
+ Module snd-usb-caiaq
+ --------------------
+
+ Module for caiaq UB audio interfaces,
+ * Native Instruments RigKontrol2
+ * Native Instruments Kore Controller
+ * Native Instruments Audio Kontrol 1
+ * Native Instruments Audio 8 DJ
+
+ This module supports multiple devices, autoprobe and hotplugging.
+
+ Module snd-usb-usx2y
+ --------------------
+
+ Module for Tascam USB US-122, US-224 and US-428 devices.
+
+ This module supports multiple devices, autoprobe and hotplugging.
+
+ Note: you need to load the firmware via usx2yloader utility included
+ in alsa-tools and alsa-firmware packages.
+
+ Module snd-via82xx
+ ------------------
+
+ Module for AC'97 motherboards based on VIA 82C686A/686B, 8233,
+ 8233A, 8233C, 8235, 8237 (south) bridge.
+
+ mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup
+ [VIA686A/686B only]
+ joystick - Enable joystick (default off) [VIA686A/686B only]
+ ac97_clock - AC'97 codec clock base (default 48000Hz)
+ dxs_support - support DXS channels,
+ 0 = auto (default), 1 = enable, 2 = disable,
+ 3 = 48k only, 4 = no VRA, 5 = enable any sample
+ rate and different sample rates on different
+ channels
+ [VIA8233/C, 8235, 8237 only]
+ ac97_quirk - AC'97 workaround for strange hardware
+ See "AC97 Quirk Option" section below.
+
+ This module supports one chip and autoprobe.
+
+ Note: on some SMP motherboards like MSI 694D the interrupts might
+ not be generated properly. In such a case, please try to
+ set the SMP (or MPS) version on BIOS to 1.1 instead of
+ default value 1.4. Then the interrupt number will be
+ assigned under 15. You might also upgrade your BIOS.
+
+ Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound)
+ channels as the first PCM. On these channels, up to 4
+ streams can be played at the same time, and the controller
+ can perform sample rate conversion with separate rates for
+ each channel.
+ As default (dxs_support = 0), 48k fixed rate is chosen
+ except for the known devices since the output is often
+ noisy except for 48k on some mother boards due to the
+ bug of BIOS.
+ Please try once dxs_support=5 and if it works on other
+ sample rates (e.g. 44.1kHz of mp3 playback), please let us
+ know the PCI subsystem vendor/device id's (output of
+ "lspci -nv").
+ If dxs_support=5 does not work, try dxs_support=4; if it
+ doesn't work too, try dxs_support=1. (dxs_support=1 is
+ usually for old motherboards. The correct implemented
+ board should work with 4 or 5.) If it still doesn't
+ work and the default setting is ok, dxs_support=3 is the
+ right choice. If the default setting doesn't work at all,
+ try dxs_support=2 to disable the DXS channels.
+ In any cases, please let us know the result and the
+ subsystem vendor/device ids. See "Links and Addresses"
+ below.
+
+ Note: for the MPU401 on VIA823x, use snd-mpu401 driver
+ additionally. The mpu_port option is for VIA686 chips only.
+
+ The power-management is supported.
+
+ Module snd-via82xx-modem
+ ------------------------
+
+ Module for VIA82xx AC97 modem
+
+ ac97_clock - AC'97 codec clock base (default 48000Hz)
+
+ This module supports one card and autoprobe.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ The power-management is supported.
+
+ Module snd-virmidi
+ ------------------
+
+ Module for virtual rawmidi devices.
+ This module creates virtual rawmidi devices which communicate
+ to the corresponding ALSA sequencer ports.
+
+ midi_devs - MIDI devices # (1-4, default=4)
+
+ This module supports multiple cards.
+
+ Module snd-virtuoso
+ -------------------
+
+ Module for sound cards based on the Asus AV66/AV100/AV200 chips,
+ i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe),
+ Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim.
+
+ This module supports autoprobe and multiple cards.
+
+ Module snd-vx222
+ ----------------
+
+ Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards.
+
+ mic - Enable Microphone on V222 Mic (NYI)
+ ibl - Capture IBL size. (default = 0, minimum size)
+
+ This module supports multiple cards.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via vxloader utility in alsa-tools package. To invoke
+ vxloader automatically, add the following to /etc/modprobe.d/alsa.conf
+
+ install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader
+
+ (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to
+ /etc/modules.conf, instead.)
+ IBL size defines the interrupts period for PCM. The smaller size
+ gives smaller latency but leads to more CPU consumption, too.
+ The size is usually aligned to 126. As default (=0), the smallest
+ size is chosen. The possible IBL values can be found in
+ /proc/asound/cardX/vx-status proc file.
+
+ The power-management is supported.
+
+ Module snd-vxpocket
+ -------------------
+
+ Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards.
+
+ ibl - Capture IBL size. (default = 0, minimum size)
+
+ This module supports multiple cards. The module is compiled only when
+ PCMCIA is supported on kernel.
+
+ With the older 2.6.x kernel, to activate the driver via the card
+ manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the
+ sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no
+ longer require a config file.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via vxloader utility in alsa-tools package.
+
+ About capture IBL, see the description of snd-vx222 module.
+
+ Note: snd-vxp440 driver is merged to snd-vxpocket driver since
+ ALSA 1.0.10.
+
+ The power-management is supported.
+
+ Module snd-ymfpci
+ -----------------
+
+ Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x).
+
+ mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default,
+ 1 (auto-detect for YMF744/754 only)
+ fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default
+ 1 (auto-detect for YMF744/754 only)
+ joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default,
+ 1 (auto-detect)
+ rear_switch - enable shared rear/line-in switch (bool)
+
+ This module supports autoprobe and multiple chips.
+
+ The power-management is supported.
+
+ Module snd-pdaudiocf
+ --------------------
+
+ Module for Sound Core PDAudioCF sound card.
+
+ The power-management is supported.
+
+
+AC97 Quirk Option
+=================
+
+The ac97_quirk option is used to enable/override the workaround for
+specific devices on drivers for on-board AC'97 controllers like
+snd-intel8x0. Some hardware have swapped output pins between Master
+and Headphone, or Surround (thanks to confusion of AC'97
+specifications from version to version :-)
+
+The driver provides the auto-detection of known problematic devices,
+but some might be unknown or wrongly detected. In such a case, pass
+the proper value with this option.
+
+The following strings are accepted:
+ - default Don't override the default setting
+ - none Disable the quirk
+ - hp_only Bind Master and Headphone controls as a single control
+ - swap_hp Swap headphone and master controls
+ - swap_surround Swap master and surround controls
+ - ad_sharing For AD1985, turn on OMS bit and use headphone
+ - alc_jack For ALC65x, turn on the jack sense mode
+ - inv_eapd Inverted EAPD implementation
+ - mute_led Bind EAPD bit for turning on/off mute LED
+
+For backward compatibility, the corresponding integer value -1, 0,
+... are accepted, too.
+
+For example, if "Master" volume control has no effect on your device
+but only "Headphone" does, pass ac97_quirk=hp_only module option.
+
+
+Configuring Non-ISAPNP Cards
+============================
+
+When the kernel is configured with ISA-PnP support, the modules
+supporting the isapnp cards will have module options "isapnp".
+If this option is set, *only* the ISA-PnP devices will be probed.
+For probing the non ISA-PnP cards, you have to pass "isapnp=0" option
+together with the proper i/o and irq configuration.
+
+When the kernel is configured without ISA-PnP support, isapnp option
+will be not built in.
+
+
+Module Autoloading Support
+==========================
+
+The ALSA drivers can be loaded automatically on demand by defining
+module aliases. The string 'snd-card-%1' is requested for ALSA native
+devices where %i is sound card number from zero to seven.
+
+To auto-load an ALSA driver for OSS services, define the string
+'sound-slot-%i' where %i means the slot number for OSS, which
+corresponds to the card index of ALSA. Usually, define this
+as the same card module.
+
+An example configuration for a single emu10k1 card is like below:
+----- /etc/modprobe.d/alsa.conf
+alias snd-card-0 snd-emu10k1
+alias sound-slot-0 snd-emu10k1
+----- /etc/modprobe.d/alsa.conf
+
+The available number of auto-loaded sound cards depends on the module
+option "cards_limit" of snd module. As default it's set to 1.
+To enable the auto-loading of multiple cards, specify the number of
+sound cards in that option.
+
+When multiple cards are available, it'd better to specify the index
+number for each card via module option, too, so that the order of
+cards is kept consistent.
+
+An example configuration for two sound cards is like below:
+
+----- /etc/modprobe.d/alsa.conf
+# ALSA portion
+options snd cards_limit=2
+alias snd-card-0 snd-interwave
+alias snd-card-1 snd-ens1371
+options snd-interwave index=0
+options snd-ens1371 index=1
+# OSS/Free portion
+alias sound-slot-0 snd-interwave
+alias sound-slot-1 snd-ens1371
+----- /etc/modprobe.d/alsa.conf
+
+In this example, the interwave card is always loaded as the first card
+(index 0) and ens1371 as the second (index 1).
+
+Alternative (and new) way to fixate the slot assignment is to use
+"slots" option of snd module. In the case above, specify like the
+following:
+
+options snd slots=snd-interwave,snd-ens1371
+
+Then, the first slot (#0) is reserved for snd-interwave driver, and
+the second (#1) for snd-ens1371. You can omit index option in each
+driver if slots option is used (although you can still have them at
+the same time as long as they don't conflict).
+
+The slots option is especially useful for avoiding the possible
+hot-plugging and the resultant slot conflict. For example, in the
+case above again, the first two slots are already reserved. If any
+other driver (e.g. snd-usb-audio) is loaded before snd-interwave or
+snd-ens1371, it will be assigned to the third or later slot.
+
+When a module name is given with '!', the slot will be given for any
+modules but that name. For example, "slots=!snd-pcsp" will reserve
+the first slot for any modules but snd-pcsp.
+
+
+ALSA PCM devices to OSS devices mapping
+=======================================
+
+/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4
+/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3
+/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12
+/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20
+/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19
+/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28
+/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36
+/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39
+/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44
+
+The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means
+sound card number and second means device number. The ALSA devices
+have either 'c' or 'p' suffix indicating the direction, capture and
+playback, respectively.
+
+Please note that the device mapping above may be varied via the module
+options of snd-pcm-oss module.
+
+
+Proc interfaces (/proc/asound)
+==============================
+
+/proc/asound/card#/pcm#[cp]/oss
+-------------------------------
+ String "erase" - erase all additional information about OSS applications
+ String "<app_name> <fragments> <fragment_size> [<options>]"
+
+ <app_name> - name of application with (higher priority) or without path
+ <fragments> - number of fragments or zero if auto
+ <fragment_size> - size of fragment in bytes or zero if auto
+ <options> - optional parameters
+ - disable the application tries to open a pcm device for
+ this channel but does not want to use it.
+ (Cause a bug or mmap needs)
+ It's good for Quake etc...
+ - direct don't use plugins
+ - block force block mode (rvplayer)
+ - non-block force non-block mode
+ - whole-frag write only whole fragments (optimization affecting
+ playback only)
+ - no-silence do not fill silence ahead to avoid clicks
+ - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl
+ instead of filled blocks
+
+ Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss
+ echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss
+ echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss
+
+
+Early Buffer Allocation
+=======================
+
+Some drivers (e.g. hdsp) require the large contiguous buffers, and
+sometimes it's too late to find such spaces when the driver module is
+actually loaded due to memory fragmentation. You can pre-allocate the
+PCM buffers by loading snd-page-alloc module and write commands to its
+proc file in prior, for example, in the early boot stage like
+/etc/init.d/*.local scripts.
+
+Reading the proc file /proc/drivers/snd-page-alloc shows the current
+usage of page allocation. In writing, you can send the following
+commands to the snd-page-alloc driver:
+
+ - add VENDOR DEVICE MASK SIZE BUFFERS
+
+ VENDOR and DEVICE are PCI vendor and device IDs. They take
+ integer numbers (0x prefix is needed for the hex).
+ MASK is the PCI DMA mask. Pass 0 if not restricted.
+ SIZE is the size of each buffer to allocate. You can pass
+ k and m suffix for KB and MB. The max number is 16MB.
+ BUFFERS is the number of buffers to allocate. It must be greater
+ than 0. The max number is 4.
+
+ - erase
+
+ This will erase the all pre-allocated buffers which are not in
+ use.
+
+
+Links and Addresses
+===================
+
+ ALSA project homepage
+ http://www.alsa-project.org
+
+ Kernel Bugzilla
+ http://bugzilla.kernel.org/
+
+ ALSA Developers ML
+ mailto:alsa-devel@alsa-project.org
+
+ alsa-info.sh script
+ http://www.alsa-project.org/alsa-info.sh
diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
new file mode 100644
index 000000000..7f10dc6ff
--- /dev/null
+++ b/Documentation/sound/alsa/Audigy-mixer.txt
@@ -0,0 +1,345 @@
+
+ Sound Blaster Audigy mixer / default DSP code
+ ===========================================
+
+This is based on SB-Live-mixer.txt.
+
+The EMU10K2 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be wrapped
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC - digital to analog converter
+ADC - analog to digital converter
+I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
+ (this standard is used for connecting standalone DAC and ADC converters)
+LFE - low frequency effects (subwoofer signal)
+AC97 - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
+ Each of the synthesizer voices can feed its output to these accumulators
+ and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+
+This control is used to attenuate sample for LFE PCM FX-bus accumulator.
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+
+This control is used to attenuate samples for front, surround, center and
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+2) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+ 0 - mono, default 0xffff (no attenuation)
+ 1 - left, default 0xffff (no attenuation)
+ 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+ 0 - mono, A destination (FX-bus 0-63), default 0
+ 1 - mono, B destination (FX-bus 0-63), default 1
+ 2 - mono, C destination (FX-bus 0-63), default 2
+ 3 - mono, D destination (FX-bus 0-63), default 3
+ 4 - mono, E destination (FX-bus 0-63), default 0
+ 5 - mono, F destination (FX-bus 0-63), default 0
+ 6 - mono, G destination (FX-bus 0-63), default 0
+ 7 - mono, H destination (FX-bus 0-63), default 0
+ 8 - left, A destination (FX-bus 0-63), default 0
+ 9 - left, B destination (FX-bus 0-63), default 1
+ 10 - left, C destination (FX-bus 0-63), default 2
+ 11 - left, D destination (FX-bus 0-63), default 3
+ 12 - left, E destination (FX-bus 0-63), default 0
+ 13 - left, F destination (FX-bus 0-63), default 0
+ 14 - left, G destination (FX-bus 0-63), default 0
+ 15 - left, H destination (FX-bus 0-63), default 0
+ 16 - right, A destination (FX-bus 0-63), default 0
+ 17 - right, B destination (FX-bus 0-63), default 1
+ 18 - right, C destination (FX-bus 0-63), default 2
+ 19 - right, D destination (FX-bus 0-63), default 3
+ 20 - right, E destination (FX-bus 0-63), default 0
+ 21 - right, F destination (FX-bus 0-63), default 0
+ 22 - right, G destination (FX-bus 0-63), default 0
+ 23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
+more than once (it means 0=0 && 1=0 is an invalid combination).
+
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+ 0 - mono, A destination attn, default 255 (no attenuation)
+ 1 - mono, B destination attn, default 255 (no attenuation)
+ 2 - mono, C destination attn, default 0 (mute)
+ 3 - mono, D destination attn, default 0 (mute)
+ 4 - mono, E destination attn, default 0 (mute)
+ 5 - mono, F destination attn, default 0 (mute)
+ 6 - mono, G destination attn, default 0 (mute)
+ 7 - mono, H destination attn, default 0 (mute)
+ 8 - left, A destination attn, default 255 (no attenuation)
+ 9 - left, B destination attn, default 0 (mute)
+ 10 - left, C destination attn, default 0 (mute)
+ 11 - left, D destination attn, default 0 (mute)
+ 12 - left, E destination attn, default 0 (mute)
+ 13 - left, F destination attn, default 0 (mute)
+ 14 - left, G destination attn, default 0 (mute)
+ 15 - left, H destination attn, default 0 (mute)
+ 16 - right, A destination attn, default 0 (mute)
+ 17 - right, B destination attn, default 255 (no attenuation)
+ 18 - right, C destination attn, default 0 (mute)
+ 19 - right, D destination attn, default 0 (mute)
+ 20 - right, E destination attn, default 0 (mute)
+ 21 - right, F destination attn, default 0 (mute)
+ 22 - right, G destination attn, default 0 (mute)
+ 23 - right, H destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+ Files:
+ LM4545.pdf AC97 Codec
+
+ m2049.pdf The EMU10K1 Digital Audio Processor
+
+ hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+ Patent numbers:
+ WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+ streams
+
+ WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+ WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+ Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+ US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+ US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+ with a multiport memory onto which multiple asynchronous
+ digital sound samples can be concurrently loaded
+
+ US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+ US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+ US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000)
+ system bus with prioritization and modification of bus transfers
+ in accordance with loop ends and minimum block sizes
+
+ US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
+ pool of short term memory registers
+
+ US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
+ a common interrupt by associating programs to GP registers,
+ defining interrupt register, polling GP registers, and invoking
+ callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
new file mode 100644
index 000000000..e7a5ed4dc
--- /dev/null
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -0,0 +1,442 @@
+ Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5
+ ========================================================
+
+ Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with
+ALSA and JACK.
+
+History
+=======
+* v1.4 - Thibault Le Meur (2007-07-11)
+ - Added Low Endianness nature of 16bits-modes
+ found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+ - Modifying document structure
+* v1.5 - Thibault Le Meur (2007-07-12)
+ - Added AC3/DTS passthru info
+
+
+1 - Audiophile USB Specs and correct usage
+==========================================
+
+This part is a reminder of important facts about the functions and limitations
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+ * Analog Stereo Input (Ai)
+ - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA)
+ - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+ are disabled
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following characteristics:
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two interfaces can't use different sample depths at the same time.
+Moreover, the Audiophile USB documentation gives the following Warning:
+"Please exit any audio application running before switching between bit depths"
+
+Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be
+activated at the same time depending on the audio mode selected:
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
+ - Ai+Ao+Di+Do
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out,
+ or 2 channels in + 4 channels out
+ - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
+ - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+ * The Do port additionally supports surround-encoded AC-3 and DTS passthrough,
+though I haven't tested it under Linux
+ - Note that in this setup only the Do interface can be enabled
+ * Apart from recording an audio digital stream, enabling the Di port is a way
+to synchronize the device to an external sample clock
+ - As a consequence, the Di port must be enable only if an active Digital
+source is connected
+ - Enabling Di when no digital source is connected can result in a
+synchronization error (for instance sound played at an odd sample rate)
+
+
+2 - Audiophile USB MIDI support in ALSA
+=======================================
+
+The Audiophile USB MIDI ports will be automatically supported once the
+following modules have been loaded:
+ * snd-usb-audio
+ * snd-seq-midi
+
+No additional setting is required.
+
+
+3 - Audiophile USB Audio support in ALSA
+========================================
+
+Audio functions of the Audiophile USB device are handled by the snd-usb-audio
+module. This module can work in a default mode (without any device-specific
+parameter), or in an "advanced" mode with the device-specific parameter called
+"device_setup".
+
+3.1 - Default Alsa driver mode
+------------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device
+capabilities at startup and activate the required mode when required
+by the applications: for instance if the user is recording in a
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample
+rates/depths automatically according to the user's needs. However, those who
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now. In this case the
+Audiophile interfaces are mapped to alsa pcm devices in the following
+way (I suppose the device's index is 1):
+ * hw:1,0 is Ao in playback and Di in capture
+ * hw:1,1 is Do in playback and Ai in capture
+ * hw:1,2 is Do in AC3/DTS passthrough mode
+
+In this mode, the device uses Big Endian byte-encoding so that
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface
+is reported to be big endian in this default driver mode.
+
+Examples:
+ * playing a S24_3BE encoded raw file to the Ao port
+ % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a S24_3BE encoded raw file from the Ai port
+ % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * playing a S16_BE encoded raw file to the Do port
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+ * playing an ac3 sample file to the Do port
+ % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
+
+If you're happy with the default Alsa driver mode and don't experience any
+issue with this mode, then you can skip the following chapter.
+
+3.2 - Advanced module setup
+---------------------------
+
+Due to the hardware constraints described above, the device initialization made
+by the Alsa driver in default mode may result in a corrupted state of the
+device. For instance, a particularly annoying issue is that the sound captured
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
+
+For people having this problem, the snd-usb-audio module has a new module
+parameter called "device_setup" (this parameter was introduced in kernel
+release 2.6.17)
+
+3.2.1 - Initializing the working mode of the Audiophile USB
+
+As far as the Audiophile USB device is concerned, this value let the user
+specify:
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not
+
+When initialized with "device_setup=0x00", the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+3.2.1.1 - 16-bit modes
+
+The two supported modes are:
+
+ * device_setup=0x01
+ - 16bits 48kHz mode with Di disabled
+ - Ai,Ao,Do can be used at the same time
+ - hw:1,0 is not available in capture mode
+ - hw:1,2 is not available
+
+ * device_setup=0x11
+ - 16bits 48kHz mode with Di enabled
+ - Ai,Ao,Di,Do can be used at the same time
+ - hw:1,0 is available in capture mode
+ - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+3.2.1.2 - 24-bit modes
+
+The three supported modes are:
+
+ * device_setup=0x09
+ - 24bits 48kHz mode with Di disabled
+ - Ai,Ao,Do can be used at the same time
+ - hw:1,0 is not available in capture mode
+ - hw:1,2 is not available
+
+ * device_setup=0x19
+ - 24bits 48kHz mode with Di enabled
+ - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
+ - hw:1,0 is available in capture mode and an active digital source must be
+ connected to Di
+ - hw:1,2 is not available
+
+ * device_setup=0x0D or 0x10
+ - 24bits 96kHz mode
+ - Di is enabled by default for this mode but does not need to be connected
+ to an active source
+ - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
+ - hw:1,0 is available in captured mode
+ - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver
+mode" above for an aplay command example)
+
+3.2.1.3 - AC3 w/ DTS passthru mode
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
+ * device_setup=0x03
+ - 16bits 48kHz mode with only the Do port enabled
+ - AC3 with DTS passthru
+ - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+ % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+3.2.2 - How to use the device_setup parameter
+----------------------------------------------
+
+The parameter can be given:
+
+ * By manually probing the device (as root):
+ # modprobe -r snd-usb-audio
+ # modprobe snd-usb-audio index=1 device_setup=0x09
+
+ * Or while configuring the modules options in your modules configuration file
+ (typically a .conf file in /etc/modprobe.d/ directory:
+ alias snd-card-1 snd-usb-audio
+ options snd-usb-audio index=1 device_setup=0x09
+
+CAUTION when initializing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+ the module BEFORE the device is turned on. So, if you use the "manual probing"
+ method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead to a misconfiguration of the device. In this case
+ turn off the device, unprobe the snd-usb-audio module, then probe it again with
+ correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+ to another mode (possibly with another sample-depth), please use also the following
+ procedure:
+ - first turn off the device
+ - de-register the snd-usb-audio module (modprobe -r)
+ - change the device_setup parameter by changing the device_setup
+ option in /etc/modprobe.d/*.conf
+ - turn on the device
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+ be enough to ensure the 'stability' of the device initialization.
+
+3.2.3 - Technical details for hackers
+-------------------------------------
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+3.2.3.1 - Audiophile USB's device_setup structure
+
+If you want to understand the device_setup magic numbers for the Audiophile
+USB, you need some very basic understanding of binary computation. However,
+this is not required to use the parameter and you may skip this section.
+
+The device_setup is one byte long and its structure is the following:
+
+ +---+---+---+---+---+---+---+---+
+ | b7| b6| b5| b4| b3| b2| b1| b0|
+ +---+---+---+---+---+---+---+---+
+ | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+ +---+---+---+---+---+---+---+---+
+
+Where:
+ * b0 is the "SET" bit
+ - it MUST be set if device_setup is initialized
+ * b1 is the "DTS" bit
+ - it is set only for Digital output with DTS/AC3
+ - this setup is not tested
+ * b2 is the Rate selection flag
+ - When set to "1" the rate range is 48.1-96kHz
+ - Otherwise the sample rate range is 8-48kHz
+ * b3 is the bit depth selection flag
+ - When set to "1" samples are 24bits long
+ - Otherwise they are 16bits long
+ - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits
+ samples
+ * b4 is the Digital input flag
+ - When set to "1" the device assumes that an active digital source is
+ connected
+ - You shouldn't enable Di if no source is seen on the port (this leads to
+ synchronization issues)
+ - b4 is implied by b2 (since only one port is enabled at a time no synch
+ error can occur)
+ * b5 to b7 are reserved for future uses, and must be set to "0"
+ - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+ * there is no check on the value you will give to device_setup
+ - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since
+ b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+ * Hardware constraints due to the USB bus limitation aren't checked
+ - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
+ only be able to use one at the same time
+
+3.2.3.2 - USB implementation details for this device
+
+You may safely skip this section if you're not interested in driver
+hacking.
+
+This section describes some internal aspects of the device and summarizes the
+data I got by usb-snooping the windows and Linux drivers.
+
+The M-Audio Audiophile USB has 7 USB Interfaces:
+a "USB interface":
+ * USB Interface nb.0
+ * USB Interface nb.1
+ - Audio Control function
+ * USB Interface nb.2
+ - Analog Output
+ * USB Interface nb.3
+ - Digital Output
+ * USB Interface nb.4
+ - Analog Input
+ * USB Interface nb.5
+ - Digital Input
+ * USB Interface nb.6
+ - MIDI interface compliant with the MIDIMAN quirk
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+ * Interface 3 (Digital Out) has an extra Alset nb.6
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5
+
+Here is a short description of the AltSettings capabilities:
+ * AltSettings 1 corresponds to
+ - 24-bit depth, 48.1-96kHz sample mode
+ - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+ * AltSettings 2 corresponds to
+ - 24-bit depth, 8-48kHz sample mode
+ - Asynch capture and playback (Ao,Ai,Do,Di)
+ * AltSettings 3 corresponds to
+ - 24-bit depth, 8-48kHz sample mode
+ - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 4 corresponds to
+ - 16-bit depth, 8-48kHz sample mode
+ - Asynch capture and playback (Ao,Ai,Do,Di)
+ * AltSettings 5 corresponds to
+ - 16-bit depth, 8-48kHz sample mode
+ - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 6 corresponds to
+ - 16-bit depth, 8-48kHz sample mode
+ - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct initialization of the device, the driver
+_must_know_ how the device will be used:
+ * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
+ registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+ Digital input is connected, and only AltSet nb.3 if Digital input
+ is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+ Digital input is connected, and only AltSet nb.5 if Digital input
+ is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the
+parse_audio_endpoints function uses a quirk called
+"audiophile_skip_setting_quirk" in order to prevent AltSettings not
+corresponding to device_setup from being registered in the driver.
+
+4 - Audiophile USB and Jack support
+===================================
+
+This section deals with support of the Audiophile USB device in Jack.
+
+There are 2 main potential issues when using Jackd with the device:
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+4.1 - Direct support in Jackd
+-----------------------------
+
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember
+exactly when this support was released into jackd, let's just say that
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting
+16bits Big-Endian devices, but since you've read carefully the above
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+ % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+4.2 - Using Alsa plughw
+-----------------------
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa "plug" converter.
+
+For instance here is one way to run Jack with 2 playback channels on Ao and 2
+capture channels from Ai:
+ % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
+
+However you may see the following warning message:
+"You appear to be using the ALSA software "plug" layer, probably a result of
+using the "default" ALSA device. This is less efficient than it could be.
+Consider using a hardware device instead rather than using the plug layer."
+
+4.3 - Getting 2 input and/or output interfaces in Jack
+------------------------------------------------------
+
+As you can see, starting the Jack server this way will only enable 1 stereo
+input (Di or Ai) and 1 stereo output (Ao or Do).
+
+This is due to the following restrictions:
+* Jack can only open one capture device and one playback device at a time
+* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
+ (and optionally hw:1,2)
+
+If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
+combine the Alsa devices into one logical "complex" device.
+
+If you want to give it a try, I recommend reading the information from
+this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
+It is related to another device (ice1712) but can be adapted to suit
+the Audiophile USB.
+
+Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
+ file
+* start jackd with this device
+
+I had no success in testing this for now, if you have any success with this kind
+of setup, please drop me an email.
diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt
new file mode 100644
index 000000000..f158cde8b
--- /dev/null
+++ b/Documentation/sound/alsa/Bt87x.txt
@@ -0,0 +1,78 @@
+Intro
+=====
+
+You might have noticed that the bt878 grabber cards have actually
+_two_ PCI functions:
+
+$ lspci
+[ ... ]
+00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+[ ... ]
+
+The first does video, it is backward compatible to the bt848. The second
+does audio. snd-bt87x is a driver for the second function. It's a sound
+driver which can be used for recording sound (and _only_ recording, no
+playback). As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+Some cards do not bother to connect anything to the audio input pins of
+the chip, and some other cards use the audio function to transport MPEG
+video data, so it's quite possible that audio recording may not work
+with your card.
+
+
+Driver Status
+=============
+
+The driver is now stable. However, it doesn't know about many TV cards,
+and it refuses to load for cards it doesn't know.
+
+If the driver complains ("Unknown TV card found, the audio driver will
+not load"), you can specify the load_all=1 option to force the driver to
+try to use the audio capture function of your card. If the frequency of
+recorded data is not right, try to specify the digital_rate option with
+other values than the default 32000 (often it's 44100 or 64000).
+
+If you have an unknown card, please mail the ID and board name to
+<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
new file mode 100644
index 000000000..4e36e6e80
--- /dev/null
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -0,0 +1,254 @@
+ Brief Notes on C-Media 8338/8738/8768/8770 Driver
+ =================================================
+
+ Takashi Iwai <tiwai@suse.de>
+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks. Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight differences between the two DACs:
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+ supports only S16LE.
+- The second DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled. Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+*** WARNING ***
+When "Four Channel Mode" switch is off, the output from rear speakers
+will be FULL VOLUME regardless of Master and PCM volumes.
+This might damage your audio equipment. Please disconnect speakers
+before your turn off this switch.
+*** WARNING ***
+
+[ Well.. I once got the output with correct volume (i.e. same with the
+ front one) and was so excited. It was even with "Four Channel" bit
+ on and "double DAC" mode. Actually I could hear separate 4 channels
+ from front and rear speakers! But.. after reboot, all was gone.
+ It's a very pity that I didn't save the register dump at that
+ time.. Maybe there is an unknown register to achieve this... ]
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default. If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else. When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function. This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6". You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
+4) channels. While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels. Since the multi-channel playback mode uses both DACs, you
+cannot operate with full-duplex.
+
+The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
+in alsa-lib. For example, you can play a WAV file with 6 channels like
+
+ % aplay -Dsurround51 sixchannels.wav
+
+For programming the 4/6 channel playback, you need to specify the PCM
+channels as you like and set the format S16LE. For example, for playback
+with 4 channels,
+
+ snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+ // or mmap if you like
+ snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+ snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+and use the interleaved 4 channel data.
+
+There are some control switches affecting to the speaker connections:
+
+"Line-In Mode" - an enum control to change the behavior of line-in
+ jack. Either "Line-In", "Rear Output" or "Bass Output" can
+ be selected. The last item is available only with model 039
+ or newer.
+ When "Rear Output" is chosen, the surround channels 3 and 4
+ are output to line-in jack.
+"Mic-In Mode" - an enum control to change the behavior of mic-in
+ jack. Either "Mic-In" or "Center/LFE Output" can be
+ selected.
+ When "Center/LFE Output" is chosen, the center and bass
+ channels (channels 5 and 6) are output to mic-in jack.
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very cheap
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2). Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+
+ % aplay -Dhw:0,2 foo.wav
+
+or
+
+ % aplay -Dspdif foo.wav
+
+24bit format is also supported experimentally.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+respectively, so you cannot playback both analog and digital streams
+simultaneously.
+
+To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+control via mixer or alsactl ("IEC958" is the official name of
+so-called S/PDIF). Then you'll see the red light on from the card so
+you know that's working obviously :)
+The SPDIF input is always enabled, so you can hear SPDIF input data
+from line-out with "IEC958 In Monitor" switch at any time (see
+below).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used. Otherwise it's turned
+off. (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
+ output through SPDIF. This switch appears only on old chip
+ models (CM8738 033 and 037).
+ Note: without this control you can output PCM to SPDIF.
+ This is "mixing" of streams, so e.g. it's not for AC3 output
+ (see the next section).
+
+"IEC958 In Select" - Select SPDIF input, the internal CD-in (false)
+ and the external input (true).
+
+"IEC958 Loop" - SPDIF input data is loop back into SPDIF
+ output (aka bypass)
+
+"IEC958 Copyright" - Set the copyright bit.
+
+"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface.
+ On some cards this doesn't work and you need to change the
+ configuration with hardware dip-switch.
+
+"IEC958 In Monitor" - SPDIF input is routed to DAC.
+
+"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
+ [FIXME: this doesn't work on all chips..]
+
+"IEC958 In Valid" - Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF. This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib. See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop". The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+On the model 033, AC3 is implemented by the software conversion in
+the alsa-lib. If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option. This doesn't matter
+on the newer models.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes. Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+MIDI CONTROLLER
+---------------
+
+With CMI8338 chips, the MPU401-UART interface is disabled as default.
+You need to set the module option "mpu_port" to a valid I/O port address
+to enable MIDI support. Valid I/O ports are 0x300, 0x310, 0x320 and
+0x330. Choose a value that doesn't conflict with other cards.
+
+With CMI8738 and newer chips, the MIDI interface is enabled by default
+and the driver automatically chooses a port address.
+
+There is _no_ hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation. On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+I don't know why..
+
+CMI8768 and newer chips do not have the FM synth.
+
+
+Joystick and Modem
+------------------
+
+The legacy joystick is supported. To enable the joystick support, pass
+joystick_port=1 module option. The value 1 means the auto-detection.
+If the auto-detection fails, try to pass the exact I/O address.
+
+The modem is enabled dynamically via a card control switch "Modem".
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci. If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt
new file mode 100644
index 000000000..3c43d1a4c
--- /dev/null
+++ b/Documentation/sound/alsa/Channel-Mapping-API.txt
@@ -0,0 +1,153 @@
+ALSA PCM channel-mapping API
+============================
+ Takashi Iwai <tiwai@suse.de>
+
+GENERAL
+-------
+
+The channel mapping API allows user to query the possible channel maps
+and the current channel map, also optionally to modify the channel map
+of the current stream.
+
+A channel map is an array of position for each PCM channel.
+Typically, a stereo PCM stream has a channel map of
+ { front_left, front_right }
+while a 4.0 surround PCM stream has a channel map of
+ { front left, front right, rear left, rear right }.
+
+The problem, so far, was that we had no standard channel map
+explicitly, and applications had no way to know which channel
+corresponds to which (speaker) position. Thus, applications applied
+wrong channels for 5.1 outputs, and you hear suddenly strange sound
+from rear. Or, some devices secretly assume that center/LFE is the
+third/fourth channels while others that C/LFE as 5th/6th channels.
+
+Also, some devices such as HDMI are configurable for different speaker
+positions even with the same number of total channels. However, there
+was no way to specify this because of lack of channel map
+specification. These are the main motivations for the new channel
+mapping API.
+
+
+DESIGN
+------
+
+Actually, "the channel mapping API" doesn't introduce anything new in
+the kernel/user-space ABI perspective. It uses only the existing
+control element features.
+
+As a ground design, each PCM substream may contain a control element
+providing the channel mapping information and configuration. This
+element is specified by:
+ iface = SNDRV_CTL_ELEM_IFACE_PCM
+ name = "Playback Channel Map" or "Capture Channel Map"
+ device = the same device number for the assigned PCM substream
+ index = the same index number for the assigned PCM substream
+
+Note the name is different depending on the PCM substream direction.
+
+Each control element provides at least the TLV read operation and the
+read operation. Optionally, the write operation can be provided to
+allow user to change the channel map dynamically.
+
+* TLV
+
+The TLV operation gives the list of available channel
+maps. A list item of a channel map is usually a TLV of
+ type data-bytes ch0 ch1 ch2...
+where type is the TLV type value, the second argument is the total
+bytes (not the numbers) of channel values, and the rest are the
+position value for each channel.
+
+As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED,
+SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used.
+The _FIXED type is for a channel map with the fixed channel position
+while the latter two are for flexible channel positions. _VAR type is
+for a channel map where all channels are freely swappable and _PAIRED
+type is where pair-wise channels are swappable. For example, when you
+have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap
+only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and
+RR.
+
+These new TLV types are defined in sound/tlv.h.
+
+The available channel position values are defined in sound/asound.h,
+here is a cut:
+
+/* channel positions */
+enum {
+ SNDRV_CHMAP_UNKNOWN = 0,
+ SNDRV_CHMAP_NA, /* N/A, silent */
+ SNDRV_CHMAP_MONO, /* mono stream */
+ /* this follows the alsa-lib mixer channel value + 3 */
+ SNDRV_CHMAP_FL, /* front left */
+ SNDRV_CHMAP_FR, /* front right */
+ SNDRV_CHMAP_RL, /* rear left */
+ SNDRV_CHMAP_RR, /* rear right */
+ SNDRV_CHMAP_FC, /* front center */
+ SNDRV_CHMAP_LFE, /* LFE */
+ SNDRV_CHMAP_SL, /* side left */
+ SNDRV_CHMAP_SR, /* side right */
+ SNDRV_CHMAP_RC, /* rear center */
+ /* new definitions */
+ SNDRV_CHMAP_FLC, /* front left center */
+ SNDRV_CHMAP_FRC, /* front right center */
+ SNDRV_CHMAP_RLC, /* rear left center */
+ SNDRV_CHMAP_RRC, /* rear right center */
+ SNDRV_CHMAP_FLW, /* front left wide */
+ SNDRV_CHMAP_FRW, /* front right wide */
+ SNDRV_CHMAP_FLH, /* front left high */
+ SNDRV_CHMAP_FCH, /* front center high */
+ SNDRV_CHMAP_FRH, /* front right high */
+ SNDRV_CHMAP_TC, /* top center */
+ SNDRV_CHMAP_TFL, /* top front left */
+ SNDRV_CHMAP_TFR, /* top front right */
+ SNDRV_CHMAP_TFC, /* top front center */
+ SNDRV_CHMAP_TRL, /* top rear left */
+ SNDRV_CHMAP_TRR, /* top rear right */
+ SNDRV_CHMAP_TRC, /* top rear center */
+ SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
+};
+
+When a PCM stream can provide more than one channel map, you can
+provide multiple channel maps in a TLV container type. The TLV data
+to be returned will contain such as:
+ SNDRV_CTL_TLVT_CONTAINER 96
+ SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC
+ SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR
+ SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \
+ SNDRV_CHMAP_RL SNDRV_CHMAP_RR
+
+The channel position is provided in LSB 16bits. The upper bits are
+used for bit flags.
+
+#define SNDRV_CHMAP_POSITION_MASK 0xffff
+#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16)
+#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16)
+
+SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted,
+(thus summing left and right channels would result in almost silence).
+Some digital mic devices have this.
+
+When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values
+don't follow the standard definition above but driver-specific.
+
+* READ OPERATION
+
+The control read operation is for providing the current channel map of
+the given stream. The control element returns an integer array
+containing the position of each channel.
+
+When this is performed before the number of the channel is specified
+(i.e. hw_params is set), it should return all channels set to
+UNKNOWN.
+
+* WRITE OPERATION
+
+The control write operation is optional, and only for devices that can
+change the channel configuration on the fly, such as HDMI. User needs
+to pass an integer value containing the valid channel positions for
+all channels of the assigned PCM substream.
+
+This operation is allowed only at PCM PREPARED state. When called in
+other states, it shall return an error.
diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt
new file mode 100644
index 000000000..3fc1cf50d
--- /dev/null
+++ b/Documentation/sound/alsa/ControlNames.txt
@@ -0,0 +1,107 @@
+This document describes standard names of mixer controls.
+
+Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
+
+DIRECTION:
+ <nothing> (both directions)
+ Playback
+ Capture
+ Bypass Playback
+ Bypass Capture
+
+FUNCTION:
+ Switch (on/off switch)
+ Volume
+ Route (route control, hardware specific)
+
+CHANNEL:
+ <nothing> (channel independent, or applies to all channels)
+ Front
+ Surround (rear left/right in 4.0/5.1 surround)
+ CLFE
+ Center
+ LFE
+ Side (side left/right for 7.1 surround)
+
+LOCATION: (physical location of source)
+ Front
+ Rear
+ Dock (docking station)
+ Internal
+
+SOURCE:
+ Master
+ Master Mono
+ Hardware Master
+ Speaker (internal speaker)
+ Bass Speaker (internal LFE speaker)
+ Headphone
+ Line Out
+ Beep (beep generator)
+ Phone
+ Phone Input
+ Phone Output
+ Synth
+ FM
+ Mic
+ Headset Mic (mic part of combined headset jack - 4-pin headphone + mic)
+ Headphone Mic (mic part of either/or - 3-pin headphone or mic)
+ Line (input only, use "Line Out" for output)
+ CD
+ Video
+ Zoom Video
+ Aux
+ PCM
+ PCM Pan
+ Loopback
+ Analog Loopback (D/A -> A/D loopback)
+ Digital Loopback (playback -> capture loopback - without analog path)
+ Mono
+ Mono Output
+ Multi
+ ADC
+ Wave
+ Music
+ I2S
+ IEC958
+ HDMI
+ SPDIF (output only)
+ SPDIF In
+ Digital In
+ HDMI/DP (either HDMI or DisplayPort)
+
+Exceptions (deprecated):
+ [Analogue|Digital] Capture Source
+ [Analogue|Digital] Capture Switch (aka input gain switch)
+ [Analogue|Digital] Capture Volume (aka input gain volume)
+ [Analogue|Digital] Playback Switch (aka output gain switch)
+ [Analogue|Digital] Playback Volume (aka output gain volume)
+ Tone Control - Switch
+ Tone Control - Bass
+ Tone Control - Treble
+ 3D Control - Switch
+ 3D Control - Center
+ 3D Control - Depth
+ 3D Control - Wide
+ 3D Control - Space
+ 3D Control - Level
+ Mic Boost [(?dB)]
+
+PCM interface:
+
+ Sample Clock Source { "Word", "Internal", "AutoSync" }
+ Clock Sync Status { "Lock", "Sync", "No Lock" }
+ External Rate /* external capture rate */
+ Capture Rate /* capture rate taken from external source */
+
+IEC958 (S/PDIF) interface:
+
+ IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */
+ IEC958 [...] [Playback|Capture] Volume /* digital volume control */
+ IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */
+ IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */
+ IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */
+ IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */
+ IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */
+ IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */
+ IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */
diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
new file mode 100644
index 000000000..e9621e349
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -0,0 +1,116 @@
+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
new file mode 100644
index 000000000..5a3163cac
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -0,0 +1,314 @@
+ 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
+======
+ inv-dmic Inverted internal mic workaround
+
+ALC267/268
+==========
+ inv-dmic Inverted internal mic workaround
+
+ALC269/270/275/276/28x/29x
+======
+ laptop-amic Laptops with analog-mic input
+ laptop-dmic Laptops with digital-mic input
+ alc269-dmic Enable ALC269(VA) digital mic workaround
+ alc271-dmic Enable ALC271X digital mic workaround
+ inv-dmic Inverted internal mic workaround
+ headset-mic Indicates a combined headset (headphone+mic) jack
+ lenovo-dock Enables docking station I/O for some Lenovos
+ dell-headset-multi Headset jack, which can also be used as mic-in
+ dell-headset-dock Headset jack (without mic-in), and also dock I/O
+
+ALC66x/67x/892
+==============
+ mario Chromebook mario model fixup
+ asus-mode1 ASUS
+ asus-mode2 ASUS
+ asus-mode3 ASUS
+ asus-mode4 ASUS
+ asus-mode5 ASUS
+ asus-mode6 ASUS
+ asus-mode7 ASUS
+ asus-mode8 ASUS
+ inv-dmic Inverted internal mic workaround
+ dell-headset-multi Headset jack, which can also be used as mic-in
+
+ALC680
+======
+ N/A
+
+ALC88x/898/1150
+======================
+ acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G
+ acer-aspire-8930g Acer Aspire 8330G/6935G
+ acer-aspire Acer Aspire others
+ inv-dmic Inverted internal mic workaround
+ no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
+
+ALC861/660
+==========
+ N/A
+
+ALC861VD/660VD
+==============
+ N/A
+
+CMI9880
+=======
+ minimal 3-jack in back
+ min_fp 3-jack in back, 2-jack in front
+ full 6-jack in back, 2-jack in front
+ full_dig 6-jack in back, 2-jack in front, SPDIF I/O
+ allout 5-jack in back, 2-jack in front, SPDIF out
+ auto auto-config reading BIOS (default)
+
+AD1882 / AD1882A
+================
+ 3stack 3-stack mode
+ 3stack-automute 3-stack with automute front HP (default)
+ 6stack 6-stack mode
+
+AD1884A / AD1883 / AD1984A / AD1984B
+====================================
+ desktop 3-stack desktop (default)
+ laptop laptop with HP jack sensing
+ mobile mobile devices with HP jack sensing
+ thinkpad Lenovo Thinkpad X300
+ touchsmart HP Touchsmart
+
+AD1884
+======
+ N/A
+
+AD1981
+======
+ basic 3-jack (default)
+ hp HP nx6320
+ thinkpad Lenovo Thinkpad T60/X60/Z60
+ toshiba Toshiba U205
+
+AD1983
+======
+ N/A
+
+AD1984
+======
+ basic default configuration
+ thinkpad Lenovo Thinkpad T61/X61
+ dell_desktop Dell T3400
+
+AD1986A
+=======
+ 3stack 3-stack, shared surrounds
+ laptop 2-channel only (FSC V2060, Samsung M50)
+ laptop-imic 2-channel with built-in mic
+ eapd Turn on EAPD constantly
+
+AD1988/AD1988B/AD1989A/AD1989B
+==============================
+ 6stack 6-jack
+ 6stack-dig ditto with SPDIF
+ 3stack 3-jack
+ 3stack-dig ditto with SPDIF
+ laptop 3-jack with hp-jack automute
+ laptop-dig ditto with SPDIF
+ auto auto-config reading BIOS (default)
+
+Conexant 5045
+=============
+ laptop-hpsense Laptop with HP sense (old model laptop)
+ laptop-micsense Laptop with Mic sense (old model fujitsu)
+ laptop-hpmicsense Laptop with HP and Mic senses
+ benq Benq R55E
+ laptop-hp530 HP 530 laptop
+ test for testing/debugging purpose, almost all controls
+ can be adjusted. Appearing only when compiled with
+ $CONFIG_SND_DEBUG=y
+
+Conexant 5047
+=============
+ laptop Basic Laptop config
+ laptop-hp Laptop config for some HP models (subdevice 30A5)
+ laptop-eapd Laptop config with EAPD support
+ test for testing/debugging purpose, almost all controls
+ can be adjusted. Appearing only when compiled with
+ $CONFIG_SND_DEBUG=y
+
+Conexant 5051
+=============
+ laptop Basic Laptop config (default)
+ hp HP Spartan laptop
+ hp-dv6736 HP dv6736
+ hp-f700 HP Compaq Presario F700
+ ideapad Lenovo IdeaPad laptop
+ toshiba Toshiba Satellite M300
+
+Conexant 5066
+=============
+ laptop Basic Laptop config (default)
+ hp-laptop HP laptops, e g G60
+ asus Asus K52JU, Lenovo G560
+ dell-laptop Dell laptops
+ dell-vostro Dell Vostro
+ olpc-xo-1_5 OLPC XO 1.5
+ ideapad Lenovo IdeaPad U150
+ thinkpad Lenovo Thinkpad
+
+STAC9200
+========
+ ref Reference board
+ oqo OQO Model 2
+ dell-d21 Dell (unknown)
+ dell-d22 Dell (unknown)
+ dell-d23 Dell (unknown)
+ dell-m21 Dell Inspiron 630m, Dell Inspiron 640m
+ dell-m22 Dell Latitude D620, Dell Latitude D820
+ dell-m23 Dell XPS M1710, Dell Precision M90
+ dell-m24 Dell Latitude 120L
+ dell-m25 Dell Inspiron E1505n
+ dell-m26 Dell Inspiron 1501
+ dell-m27 Dell Inspiron E1705/9400
+ gateway-m4 Gateway laptops with EAPD control
+ gateway-m4-2 Gateway laptops with EAPD control
+ panasonic Panasonic CF-74
+ auto BIOS setup (default)
+
+STAC9205/9254
+=============
+ ref Reference board
+ dell-m42 Dell (unknown)
+ dell-m43 Dell Precision
+ dell-m44 Dell Inspiron
+ eapd Keep EAPD on (e.g. Gateway T1616)
+ auto BIOS setup (default)
+
+STAC9220/9221
+=============
+ ref Reference board
+ 3stack D945 3stack
+ 5stack D945 5stack + SPDIF
+ intel-mac-v1 Intel Mac Type 1
+ intel-mac-v2 Intel Mac Type 2
+ intel-mac-v3 Intel Mac Type 3
+ intel-mac-v4 Intel Mac Type 4
+ intel-mac-v5 Intel Mac Type 5
+ intel-mac-auto Intel Mac (detect type according to subsystem id)
+ macmini Intel Mac Mini (equivalent with type 3)
+ macbook Intel Mac Book (eq. type 5)
+ macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+ macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3)
+ imac-intel Intel iMac (eq. type 2)
+ imac-intel-20 Intel iMac (newer version) (eq. type 3)
+ ecs202 ECS/PC chips
+ dell-d81 Dell (unknown)
+ dell-d82 Dell (unknown)
+ dell-m81 Dell (unknown)
+ dell-m82 Dell XPS M1210
+ auto BIOS setup (default)
+
+STAC9202/9250/9251
+==================
+ ref Reference board, base config
+ m1 Some Gateway MX series laptops (NX560XL)
+ m1-2 Some Gateway MX series laptops (MX6453)
+ m2 Some Gateway MX series laptops (M255)
+ m2-2 Some Gateway MX series laptops
+ m3 Some Gateway MX series laptops
+ m5 Some Gateway MX series laptops (MP6954)
+ m6 Some Gateway NX series laptops
+ auto BIOS setup (default)
+
+STAC9227/9228/9229/927x
+=======================
+ ref Reference board
+ ref-no-jd Reference board without HP/Mic jack detection
+ 3stack D965 3stack
+ 5stack D965 5stack + SPDIF
+ 5stack-no-fp D965 5stack without front panel
+ dell-3stack Dell Dimension E520
+ dell-bios Fixes with Dell BIOS setup
+ dell-bios-amic Fixes with Dell BIOS setup including analog mic
+ volknob Fixes with volume-knob widget 0x24
+ auto BIOS setup (default)
+
+STAC92HD71B*
+============
+ ref Reference board
+ dell-m4-1 Dell desktops
+ dell-m4-2 Dell desktops
+ dell-m4-3 Dell desktops
+ hp-m4 HP mini 1000
+ hp-dv5 HP dv series
+ hp-hdx HP HDX series
+ hp-dv4-1222nr HP dv4-1222nr (with LED support)
+ auto BIOS setup (default)
+
+STAC92HD73*
+===========
+ ref Reference board
+ no-jd BIOS setup but without jack-detection
+ intel Intel DG45* mobos
+ dell-m6-amic Dell desktops/laptops with analog mics
+ dell-m6-dmic Dell desktops/laptops with digital mics
+ dell-m6 Dell desktops/laptops with both type of mics
+ dell-eq Dell desktops/laptops
+ alienware Alienware M17x
+ auto BIOS setup (default)
+
+STAC92HD83*
+===========
+ ref Reference board
+ mic-ref Reference board with power management for ports
+ dell-s14 Dell laptop
+ dell-vostro-3500 Dell Vostro 3500 laptop
+ hp-dv7-4000 HP dv-7 4000
+ hp_cNB11_intquad HP CNB models with 4 speakers
+ hp-zephyr HP Zephyr
+ hp-led HP with broken BIOS for mute LED
+ hp-inv-led HP with broken BIOS for inverted mute LED
+ hp-mic-led HP with mic-mute LED
+ headset-jack Dell Latitude with a 4-pin headset jack
+ hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f)
+ hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10)
+ hp-bnb13-eq Hardware equalizer setup for HP laptops
+ auto BIOS setup (default)
+
+STAC92HD95
+==========
+ hp-led LED support for HP laptops
+ hp-bass Bass HPF setup for HP Spectre 13
+
+STAC9872
+========
+ vaio VAIO laptop without SPDIF
+ auto BIOS setup (default)
+
+Cirrus Logic CS4206/4207
+========================
+ mbp55 MacBook Pro 5,5
+ imac27 IMac 27 Inch
+ auto BIOS setup (default)
+
+Cirrus Logic CS4208
+===================
+ mba6 MacBook Air 6,1 and 6,2
+ gpio0 Enable GPIO 0 amp
+ auto BIOS setup (default)
+
+VIA VT17xx/VT18xx/VT20xx
+========================
+ auto BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
new file mode 100644
index 000000000..e7193aac6
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -0,0 +1,863 @@
+MORE NOTES ON HD-AUDIO DRIVER
+=============================
+ Takashi Iwai <tiwai@suse.de>
+
+
+GENERAL
+-------
+
+HD-audio is the new standard on-board audio component on modern PCs
+after AC97. Although Linux has been supporting HD-audio since long
+time ago, there are often problems with new machines. A part of the
+problem is broken BIOS, and the rest is the driver implementation.
+This document explains the brief trouble-shooting and debugging
+methods for the HD-audio hardware.
+
+The HD-audio component consists of two parts: the controller chip and
+the codec chips on the HD-audio bus. Linux provides a single driver
+for all controllers, snd-hda-intel. Although the driver name contains
+a word of a well-known hardware vendor, it's not specific to it but for
+all controller chips by other companies. Since the HD-audio
+controllers are supposed to be compatible, the single snd-hda-driver
+should work in most cases. But, not surprisingly, there are known
+bugs and issues specific to each controller type. The snd-hda-intel
+driver has a bunch of workarounds for these as described below.
+
+A controller may have multiple codecs. Usually you have one audio
+codec and optionally one modem codec. In theory, there might be
+multiple audio codecs, e.g. for analog and digital outputs, and the
+driver might not work properly because of conflict of mixer elements.
+This should be fixed in future if such hardware really exists.
+
+The snd-hda-intel driver has several different codec parsers depending
+on the codec. It has a generic parser as a fallback, but this
+functionality is fairly limited until now. Instead of the generic
+parser, usually the codec-specific parser (coded in patch_*.c) is used
+for the codec-specific implementations. The details about the
+codec-specific problems are explained in the later sections.
+
+If you are interested in the deep debugging of HD-audio, read the
+HD-audio specification at first. The specification is found on
+Intel's web page, for example:
+
+- http://www.intel.com/standards/hdaudio/
+
+
+HD-AUDIO CONTROLLER
+-------------------
+
+DMA-Position Problem
+~~~~~~~~~~~~~~~~~~~~
+The most common problem of the controller is the inaccurate DMA
+pointer reporting. The DMA pointer for playback and capture can be
+read in two ways, either via a LPIB register or via a position-buffer
+map. As default the driver tries to read from the io-mapped
+position-buffer, and falls back to LPIB if the position-buffer appears
+dead. However, this detection isn't perfect on some devices. In such
+a case, you can change the default method via `position_fix` option.
+
+`position_fix=1` means to use LPIB method explicitly.
+`position_fix=2` means to use the position-buffer.
+`position_fix=3` means to use a combination of both methods, needed
+for some VIA controllers. The capture stream position is corrected
+by comparing both LPIB and position-buffer values.
+`position_fix=4` is another combination available for all controllers,
+and uses LPIB for the playback and the position-buffer for the capture
+streams.
+0 is the default value for all other
+controllers, the automatic check and fallback to LPIB as described in
+the above. If you get a problem of repeated sounds, this option might
+help.
+
+In addition to that, every controller is known to be broken regarding
+the wake-up timing. It wakes up a few samples before actually
+processing the data on the buffer. This caused a lot of problems, for
+example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts
+an artificial delay to the wake up timing. This delay is controlled
+via `bdl_pos_adj` option.
+
+When `bdl_pos_adj` is a negative value (as default), it's assigned to
+an appropriate value depending on the controller chip. For Intel
+chips, it'd be 1 while it'd be 32 for others. Usually this works.
+Only in case it doesn't work and you get warning messages, you should
+change this parameter to other values.
+
+
+Codec-Probing Problem
+~~~~~~~~~~~~~~~~~~~~~
+A less often but a more severe problem is the codec probing. When
+BIOS reports the available codec slots wrongly, the driver gets
+confused and tries to access the non-existing codec slot. This often
+results in the total screw-up, and destructs the further communication
+with the codec chips. The symptom appears usually as error messages
+like:
+------------------------------------------------------------------------
+ hda_intel: azx_get_response timeout, switching to polling mode:
+ last cmd=0x12345678
+ hda_intel: azx_get_response timeout, switching to single_cmd mode:
+ last cmd=0x12345678
+------------------------------------------------------------------------
+
+The first line is a warning, and this is usually relatively harmless.
+It means that the codec response isn't notified via an IRQ. The
+driver uses explicit polling method to read the response. It gives
+very slight CPU overhead, but you'd unlikely notice it.
+
+The second line is, however, a fatal error. If this happens, usually
+it means that something is really wrong. Most likely you are
+accessing a non-existing codec slot.
+
+Thus, if the second error message appears, try to narrow the probed
+codec slots via `probe_mask` option. It's a bitmask, and each bit
+corresponds to the codec slot. For example, to probe only the first
+slot, pass `probe_mask=1`. For the first and the third slots, pass
+`probe_mask=5` (where 5 = 1 | 4), and so on.
+
+Since 2.6.29 kernel, the driver has a more robust probing method, so
+this error might happen rarely, though.
+
+On a machine with a broken BIOS, sometimes you need to force the
+driver to probe the codec slots the hardware doesn't report for use.
+In such a case, turn the bit 8 (0x100) of `probe_mask` option on.
+Then the rest 8 bits are passed as the codec slots to probe
+unconditionally. For example, `probe_mask=0x103` will force to probe
+the codec slots 0 and 1 no matter what the hardware reports.
+
+
+Interrupt Handling
+~~~~~~~~~~~~~~~~~~
+HD-audio driver uses MSI as default (if available) since 2.6.33
+kernel as MSI works better on some machines, and in general, it's
+better for performance. However, Nvidia controllers showed bad
+regressions with MSI (especially in a combination with AMD chipset),
+thus we disabled MSI for them.
+
+There seem also still other devices that don't work with MSI. If you
+see a regression wrt the sound quality (stuttering, etc) or a lock-up
+in the recent kernel, try to pass `enable_msi=0` option to disable
+MSI. If it works, you can add the known bad device to the blacklist
+defined in hda_intel.c. In such a case, please report and give the
+patch back to the upstream developer.
+
+
+HD-AUDIO CODEC
+--------------
+
+Model Option
+~~~~~~~~~~~~
+The most common problem regarding the HD-audio driver is the
+unsupported codec features or the mismatched device configuration.
+Most of codec-specific code has several preset models, either to
+override the BIOS setup or to provide more comprehensive features.
+
+The driver checks PCI SSID and looks through the static configuration
+table until any matching entry is found. If you have a new machine,
+you may see a message like below:
+------------------------------------------------------------------------
+ hda_codec: ALC880: BIOS auto-probing.
+------------------------------------------------------------------------
+Meanwhile, in the earlier versions, you would see a message like:
+------------------------------------------------------------------------
+ hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
+------------------------------------------------------------------------
+Even if you see such a message, DON'T PANIC. Take a deep breath and
+keep your towel. First of all, it's an informational message, no
+warning, no error. This means that the PCI SSID of your device isn't
+listed in the known preset model (white-)list. But, this doesn't mean
+that the driver is broken. Many codec-drivers provide the automatic
+configuration mechanism based on the BIOS setup.
+
+The HD-audio codec has usually "pin" widgets, and BIOS sets the default
+configuration of each pin, which indicates the location, the
+connection type, the jack color, etc. The HD-audio driver can guess
+the right connection judging from these default configuration values.
+However -- some codec-support codes, such as patch_analog.c, don't
+support the automatic probing (yet as of 2.6.28). And, BIOS is often,
+yes, pretty often broken. It sets up wrong values and screws up the
+driver.
+
+The preset model (or recently called as "fix-up") is provided
+basically to overcome such a situation. When the matching preset
+model is found in the white-list, the driver assumes the static
+configuration of that preset with the correct pin setup, etc.
+Thus, if you have a newer machine with a slightly different PCI SSID
+(or codec SSID) from the existing one, you may have a good chance to
+re-use the same model. You can pass the `model` option to specify the
+preset model instead of PCI (and codec-) SSID look-up.
+
+What `model` option values are available depends on the codec chip.
+Check your codec chip from the codec proc file (see "Codec Proc-File"
+section below). It will show the vendor/product name of your codec
+chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file,
+the section of HD-audio driver. You can find a list of codecs
+and `model` options belonging to each codec. For example, for Realtek
+ALC262 codec chip, pass `model=ultra` for devices that are compatible
+with Samsung Q1 Ultra.
+
+Thus, the first thing you can do for any brand-new, unsupported and
+non-working HD-audio hardware is to check HD-audio codec and several
+different `model` option values. If you have any luck, some of them
+might suit with your device well.
+
+There are a few special model option values:
+- when 'nofixup' is passed, the device-specific fixups in the codec
+ parser are skipped.
+- when `generic` is passed, the codec-specific parser is skipped and
+ only the generic parser is used.
+
+
+Speaker and Headphone Output
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+One of the most frequent (and obvious) bugs with HD-audio is the
+silent output from either or both of a built-in speaker and a
+headphone jack. In general, you should try a headphone output at
+first. A speaker output often requires more additional controls like
+the external amplifier bits. Thus a headphone output has a slightly
+better chance.
+
+Before making a bug report, double-check whether the mixer is set up
+correctly. The recent version of snd-hda-intel driver provides mostly
+"Master" volume control as well as "Front" volume (where Front
+indicates the front-channels). In addition, there can be individual
+"Headphone" and "Speaker" controls.
+
+Ditto for the speaker output. There can be "External Amplifier"
+switch on some codecs. Turn on this if present.
+
+Another related problem is the automatic mute of speaker output by
+headphone plugging. This feature is implemented in most cases, but
+not on every preset model or codec-support code.
+
+In anyway, try a different model option if you have such a problem.
+Some other models may match better and give you more matching
+functionality. If none of the available models works, send a bug
+report. See the bug report section for details.
+
+If you are masochistic enough to debug the driver problem, note the
+following:
+
+- The speaker (and the headphone, too) output often requires the
+ external amplifier. This can be set usually via EAPD verb or a
+ certain GPIO. If the codec pin supports EAPD, you have a better
+ chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly
+ it's either GPIO0 or GPIO1) may turn on/off EAPD.
+- Some Realtek codecs require special vendor-specific coefficients to
+ turn on the amplifier. See patch_realtek.c.
+- IDT codecs may have extra power-enable/disable controls on each
+ analog pin. See patch_sigmatel.c.
+- Very rare but some devices don't accept the pin-detection verb until
+ triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the
+ codec-communication stall. Some examples are found in
+ patch_realtek.c.
+
+
+Capture Problems
+~~~~~~~~~~~~~~~~
+The capture problems are often because of missing setups of mixers.
+Thus, before submitting a bug report, make sure that you set up the
+mixer correctly. For example, both "Capture Volume" and "Capture
+Switch" have to be set properly in addition to the right "Capture
+Source" or "Input Source" selection. Some devices have "Mic Boost"
+volume or switch.
+
+When the PCM device is opened via "default" PCM (without pulse-audio
+plugin), you'll likely have "Digital Capture Volume" control as well.
+This is provided for the extra gain/attenuation of the signal in
+software, especially for the inputs without the hardware volume
+control such as digital microphones. Unless really needed, this
+should be set to exactly 50%, corresponding to 0dB -- neither extra
+gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
+this control will have no influence, though.
+
+It's known that some codecs / devices have fairly bad analog circuits,
+and the recorded sound contains a certain DC-offset. This is no bug
+of the driver.
+
+Most of modern laptops have no analog CD-input connection. Thus, the
+recording from CD input won't work in many cases although the driver
+provides it as the capture source. Use CDDA instead.
+
+The automatic switching of the built-in and external mic per plugging
+is implemented on some codec models but not on every model. Partly
+because of my laziness but mostly lack of testers. Feel free to
+submit the improvement patch to the author.
+
+
+Direct Debugging
+~~~~~~~~~~~~~~~~
+If no model option gives you a better result, and you are a tough guy
+to fight against evil, try debugging via hitting the raw HD-audio
+codec verbs to the device. Some tools are available: hda-emu and
+hda-analyzer. The detailed description is found in the sections
+below. You'd need to enable hwdep for using these tools. See "Kernel
+Configuration" section.
+
+
+OTHER ISSUES
+------------
+
+Kernel Configuration
+~~~~~~~~~~~~~~~~~~~~
+In general, I recommend you to enable the sound debug option,
+`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not.
+This enables snd_printd() macro and others, and you'll get additional
+kernel messages at probing.
+
+In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this
+will give you far more messages. Thus turn this on only when you are
+sure to want it.
+
+Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*`
+options. Note that each of them corresponds to the codec chip, not
+the controller chip. Thus, even if lspci shows the Nvidia controller,
+you may need to choose the option for other vendors. If you are
+unsure, just select all yes.
+
+`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver.
+When this is enabled, the driver creates hardware-dependent devices
+(one per each codec), and you have a raw access to the device via
+these device files. For example, `hwC0D2` will be created for the
+codec slot #2 of the first card (#0). For debug-tools such as
+hda-verb and hda-analyzer, the hwdep device has to be enabled.
+Thus, it'd be better to turn this on always.
+
+`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the
+hwdep option above. When enabled, you'll have some sysfs files under
+the corresponding hwdep directory. See "HD-audio reconfiguration"
+section below.
+
+`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature.
+See "Power-saving" section below.
+
+
+Codec Proc-File
+~~~~~~~~~~~~~~~
+The codec proc-file is a treasure-chest for debugging HD-audio.
+It shows most of useful information of each codec widget.
+
+The proc file is located in /proc/asound/card*/codec#*, one file per
+each codec slot. You can know the codec vendor, product id and
+names, the type of each widget, capabilities and so on.
+This file, however, doesn't show the jack sensing state, so far. This
+is because the jack-sensing might be depending on the trigger state.
+
+This file will be picked up by the debug tools, and also it can be fed
+to the emulator as the primary codec information. See the debug tools
+section below.
+
+This proc file can be also used to check whether the generic parser is
+used. When the generic parser is used, the vendor/product ID name
+will appear as "Realtek ID 0262", instead of "Realtek ALC262".
+
+
+HD-Audio Reconfiguration
+~~~~~~~~~~~~~~~~~~~~~~~~
+This is an experimental feature to allow you re-configure the HD-audio
+codec dynamically without reloading the driver. The following sysfs
+files are available under each codec-hwdep device directory (e.g.
+/sys/class/sound/hwC0D0):
+
+vendor_id::
+ Shows the 32bit codec vendor-id hex number. You can change the
+ vendor-id value by writing to this file.
+subsystem_id::
+ Shows the 32bit codec subsystem-id hex number. You can change the
+ subsystem-id value by writing to this file.
+revision_id::
+ Shows the 32bit codec revision-id hex number. You can change the
+ revision-id value by writing to this file.
+afg::
+ Shows the AFG ID. This is read-only.
+mfg::
+ Shows the MFG ID. This is read-only.
+name::
+ Shows the codec name string. Can be changed by writing to this
+ file.
+modelname::
+ Shows the currently set `model` option. Can be changed by writing
+ to this file.
+init_verbs::
+ The extra verbs to execute at initialization. You can add a verb by
+ writing to this file. Pass three numbers: nid, verb and parameter
+ (separated with a space).
+hints::
+ Shows / stores hint strings for codec parsers for any use.
+ Its format is `key = value`. For example, passing `jack_detect = no`
+ will disable the jack detection of the machine completely.
+init_pin_configs::
+ Shows the initial pin default config values set by BIOS.
+driver_pin_configs::
+ Shows the pin default values set by the codec parser explicitly.
+ This doesn't show all pin values but only the changed values by
+ the parser. That is, if the parser doesn't change the pin default
+ config values by itself, this will contain nothing.
+user_pin_configs::
+ Shows the pin default config values to override the BIOS setup.
+ Writing this (with two numbers, NID and value) appends the new
+ value. The given will be used instead of the initial BIOS value at
+ the next reconfiguration time. Note that this config will override
+ even the driver pin configs, too.
+reconfig::
+ Triggers the codec re-configuration. When any value is written to
+ this file, the driver re-initialize and parses the codec tree
+ again. All the changes done by the sysfs entries above are taken
+ into account.
+clear::
+ Resets the codec, removes the mixer elements and PCM stuff of the
+ specified codec, and clear all init verbs and hints.
+
+For example, when you want to change the pin default configuration
+value of the pin widget 0x14 to 0x9993013f, and let the driver
+re-configure based on that state, run like below:
+------------------------------------------------------------------------
+ # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
+ # echo 1 > /sys/class/sound/hwC0D0/reconfig
+------------------------------------------------------------------------
+
+
+Hint Strings
+~~~~~~~~~~~~
+The codec parser have several switches and adjustment knobs for
+matching better with the actual codec or device behavior. Many of
+them can be adjusted dynamically via "hints" strings as mentioned in
+the section above. For example, by passing `jack_detect = no` string
+via sysfs or a patch file, you can disable the jack detection, thus
+the codec parser will skip the features like auto-mute or mic
+auto-switch. As a boolean value, either `yes`, `no`, `true`, `false`,
+`1` or `0` can be passed.
+
+The generic parser supports the following hints:
+
+- jack_detect (bool): specify whether the jack detection is available
+ at all on this machine; default true
+- inv_jack_detect (bool): indicates that the jack detection logic is
+ inverted
+- trigger_sense (bool): indicates that the jack detection needs the
+ explicit call of AC_VERB_SET_PIN_SENSE verb
+- inv_eapd (bool): indicates that the EAPD is implemented in the
+ inverted logic
+- pcm_format_first (bool): sets the PCM format before the stream tag
+ and channel ID
+- sticky_stream (bool): keep the PCM format, stream tag and ID as long
+ as possible; default true
+- spdif_status_reset (bool): reset the SPDIF status bits at each time
+ the SPDIF stream is set up
+- pin_amp_workaround (bool): the output pin may have multiple amp
+ values
+- single_adc_amp (bool): ADCs can have only single input amps
+- auto_mute (bool): enable/disable the headphone auto-mute feature;
+ default true
+- auto_mic (bool): enable/disable the mic auto-switch feature; default
+ true
+- line_in_auto_switch (bool): enable/disable the line-in auto-switch
+ feature; default false
+- need_dac_fix (bool): limits the DACs depending on the channel count
+- primary_hp (bool): probe headphone jacks as the primary outputs;
+ default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+ line-in/surround, mic/clfe jacks)
+- multi_cap_vol (bool): provide multiple capture volumes
+- inv_dmic_split (bool): provide split internal mic volume/switch for
+ phase-inverted digital mics
+- indep_hp (bool): provide the independent headphone PCM stream and
+ the corresponding mixer control, if available
+- add_stereo_mix_input (bool): add the stereo mix (analog-loopback
+ mix) to the input mux if available
+- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each
+ I/O jack for allowing to change the headphone amp and mic bias VREF
+ capabilities
+- power_save_node (bool): advanced power management for each widget,
+ controlling the power sate (D0/D3) of each widget node depending on
+ the actual pin and stream states
+- power_down_unused (bool): power down the unused widgets, a subset of
+ power_save_node, and will be dropped in future
+- add_hp_mic (bool): add the headphone to capture source if possible
+- hp_mic_detect (bool): enable/disable the hp/mic shared input for a
+ single built-in mic case; default true
+- mixer_nid (int): specifies the widget NID of the analog-loopback
+ mixer
+
+
+Early Patching
+~~~~~~~~~~~~~~
+When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
+firmware file for modifying the HD-audio setup before initializing the
+codec. This can work basically like the reconfiguration via sysfs in
+the above, but it does it before the first codec configuration.
+
+A patch file is a plain text file which looks like below:
+
+------------------------------------------------------------------------
+ [codec]
+ 0x12345678 0xabcd1234 2
+
+ [model]
+ auto
+
+ [pincfg]
+ 0x12 0x411111f0
+
+ [verb]
+ 0x20 0x500 0x03
+ 0x20 0x400 0xff
+
+ [hint]
+ jack_detect = no
+------------------------------------------------------------------------
+
+The file needs to have a line `[codec]`. The next line should contain
+three numbers indicating the codec vendor-id (0x12345678 in the
+example), the codec subsystem-id (0xabcd1234) and the address (2) of
+the codec. The rest patch entries are applied to this specified codec
+until another codec entry is given. Passing 0 or a negative number to
+the first or the second value will make the check of the corresponding
+field be skipped. It'll be useful for really broken devices that don't
+initialize SSID properly.
+
+The `[model]` line allows to change the model name of the each codec.
+In the example above, it will be changed to model=auto.
+Note that this overrides the module option.
+
+After the `[pincfg]` line, the contents are parsed as the initial
+default pin-configurations just like `user_pin_configs` sysfs above.
+The values can be shown in user_pin_configs sysfs file, too.
+
+Similarly, the lines after `[verb]` are parsed as `init_verbs`
+sysfs entries, and the lines after `[hint]` are parsed as `hints`
+sysfs entries, respectively.
+
+Another example to override the codec vendor id from 0x12345678 to
+0xdeadbeef is like below:
+------------------------------------------------------------------------
+ [codec]
+ 0x12345678 0xabcd1234 2
+
+ [vendor_id]
+ 0xdeadbeef
+------------------------------------------------------------------------
+
+In the similar way, you can override the codec subsystem_id via
+`[subsystem_id]`, the revision id via `[revision_id]` line.
+Also, the codec chip name can be rewritten via `[chip_name]` line.
+------------------------------------------------------------------------
+ [codec]
+ 0x12345678 0xabcd1234 2
+
+ [subsystem_id]
+ 0xffff1111
+
+ [revision_id]
+ 0x10
+
+ [chip_name]
+ My-own NEWS-0002
+------------------------------------------------------------------------
+
+The hd-audio driver reads the file via request_firmware(). Thus,
+a patch file has to be located on the appropriate firmware path,
+typically, /lib/firmware. For example, when you pass the option
+`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be
+present.
+
+The patch module option is specific to each card instance, and you
+need to give one file name for each instance, separated by commas.
+For example, if you have two cards, one for an on-board analog and one
+for an HDMI video board, you may pass patch option like below:
+------------------------------------------------------------------------
+ options snd-hda-intel patch=on-board-patch,hdmi-patch
+------------------------------------------------------------------------
+
+
+Power-Saving
+~~~~~~~~~~~~
+The power-saving is a kind of auto-suspend of the device. When the
+device is inactive for a certain time, the device is automatically
+turned off to save the power. The time to go down is specified via
+`power_save` module option, and this option can be changed dynamically
+via sysfs.
+
+The power-saving won't work when the analog loopback is enabled on
+some codecs. Make sure that you mute all unneeded signal routes when
+you want the power-saving.
+
+The power-saving feature might cause audible click noises at each
+power-down/up depending on the device. Some of them might be
+solvable, but some are hard, I'm afraid. Some distros such as
+openSUSE enables the power-saving feature automatically when the power
+cable is unplugged. Thus, if you hear noises, suspect first the
+power-saving. See /sys/module/snd_hda_intel/parameters/power_save to
+check the current value. If it's non-zero, the feature is turned on.
+
+The recent kernel supports the runtime PM for the HD-audio controller
+chip, too. It means that the HD-audio controller is also powered up /
+down dynamically. The feature is enabled only for certain controller
+chips like Intel LynxPoint. You can enable/disable this feature
+forcibly by setting `power_save_controller` option, which is also
+available at /sys/module/snd_hda_intel/parameters directory.
+
+
+Tracepoints
+~~~~~~~~~~~
+The hd-audio driver gives a few basic tracepoints.
+`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
+traces the response from RIRB (only when read from the codec driver).
+`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc,
+`hda:hda_unsol_event` traces the unsolicited events, and
+`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up
+via power-saving behavior.
+
+Enabling all tracepoints can be done like
+------------------------------------------------------------------------
+ # echo 1 > /sys/kernel/debug/tracing/events/hda/enable
+------------------------------------------------------------------------
+then after some commands, you can traces from
+/sys/kernel/debug/tracing/trace file. For example, when you want to
+trace what codec command is sent, enable the tracepoint like:
+------------------------------------------------------------------------
+ # cat /sys/kernel/debug/tracing/trace
+ # tracer: nop
+ #
+ # TASK-PID CPU# TIMESTAMP FUNCTION
+ # | | | | |
+ <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
+ <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
+ <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
+ <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
+ <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
+ <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
+ <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
+ <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
+------------------------------------------------------------------------
+Here `[0:0]` indicates the card number and the codec address, and
+`val` shows the value sent to the codec, respectively. The value is
+a packed value, and you can decode it via hda-decode-verb program
+included in hda-emu package below. For example, the value e3a019 is
+to set the left output-amp value to 25.
+------------------------------------------------------------------------
+ % hda-decode-verb 0xe3a019
+ raw value = 0x00e3a019
+ cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
+ raw value: verb = 0x3a0, parm = 0x19
+ verbname = set_amp_gain_mute
+ amp raw val = 0xa019
+ output, left, idx=0, mute=0, val=25
+------------------------------------------------------------------------
+
+
+Development Tree
+~~~~~~~~~~~~~~~~
+The latest development codes for HD-audio are found on sound git tree:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+
+The master branch or for-next branches can be used as the main
+development branches in general while the development for the current
+and next kernels are found in for-linus and for-next branches,
+respectively.
+
+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 included now in alsa-tools:
+
+- git://git.alsa-project.org/alsa-tools.git
+
+Also, the old stand-alone package is found in the ftp directory:
+
+- ftp://ftp.suse.com/pub/people/tiwai/misc/
+
+Also a git repository is available:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git
+
+See README file in the tarball for more details about hda-verb
+program.
+
+
+hda-analyzer
+~~~~~~~~~~~~
+hda-analyzer provides a graphical interface to access the raw HD-audio
+control, based on pyGTK2 binding. It's a more powerful version of
+hda-verb. The program gives you an easy-to-use GUI stuff for showing
+the widget information and adjusting the amp values, as well as the
+proc-compatible output.
+
+The hda-analyzer:
+
+- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
+
+is a part of alsa.git repository in alsa-project.org:
+
+- git://git.alsa-project.org/alsa.git
+
+Codecgraph
+~~~~~~~~~~
+Codecgraph is a utility program to generate a graph and visualizes the
+codec-node connection of a codec chip. It's especially useful when
+you analyze or debug a codec without a proper datasheet. The program
+parses the given codec proc file and converts to SVG via graphiz
+program.
+
+The tarball and GIT trees are found in the web page at:
+
+- http://helllabs.org/codecgraph/
+
+
+hda-emu
+~~~~~~~
+hda-emu is an HD-audio emulator. The main purpose of this program is
+to debug an HD-audio codec without the real hardware. Thus, it
+doesn't emulate the behavior with the real audio I/O, but it just
+dumps the codec register changes and the ALSA-driver internal changes
+at probing and operating the HD-audio driver.
+
+The program requires a codec proc-file to simulate. Get a proc file
+for the target codec beforehand, or pick up an example codec from the
+codec proc collections in the tarball. Then, run the program with the
+proc file, and the hda-emu program will start parsing the codec file
+and simulates the HD-audio driver:
+
+------------------------------------------------------------------------
+ % hda-emu codecs/stac9200-dell-d820-laptop
+ # Parsing..
+ hda_codec: Unknown model for STAC9200, using BIOS defaults
+ hda_codec: pin nid 08 bios pin config 40c003fa
+ ....
+------------------------------------------------------------------------
+
+The program gives you only a very dumb command-line interface. You
+can get a proc-file dump at the current state, get a list of control
+(mixer) elements, set/get the control element value, simulate the PCM
+operation, the jack plugging simulation, etc.
+
+The 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.
+
+
+hda-jack-retask
+~~~~~~~~~~~~~~~
+hda-jack-retask is a user-friendly GUI program to manipulate the
+HD-audio pin control for jack retasking. If you have a problem about
+the jack assignment, try this program and check whether you can get
+useful results. Once when you figure out the proper pin assignment,
+it can be fixed either in the driver code statically or via passing a
+firmware patch file (see "Early Patching" section).
+
+The program is included in alsa-tools now:
+
+- git://git.alsa-project.org/alsa-tools.git
+
diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt
new file mode 100644
index 000000000..ccda41b10
--- /dev/null
+++ b/Documentation/sound/alsa/Joystick.txt
@@ -0,0 +1,86 @@
+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
new file mode 100644
index 000000000..4ee35b4fb
--- /dev/null
+++ b/Documentation/sound/alsa/MIXART.txt
@@ -0,0 +1,100 @@
+ 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
new file mode 100644
index 000000000..152ca2a3f
--- /dev/null
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -0,0 +1,305 @@
+ 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
new file mode 100644
index 000000000..7f8a0d325
--- /dev/null
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -0,0 +1,234 @@
+ Proc Files of ALSA Drivers
+ ==========================
+ Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+ALSA has its own proc tree, /proc/asound. Many useful information are
+found in this tree. When you encounter a problem and need debugging,
+check the files listed in the following sections.
+
+Each card has its subtree cardX, where X is from 0 to 7. The
+card-specific files are stored in the card* subdirectories.
+
+
+Global Information
+------------------
+
+cards
+ Shows the list of currently configured ALSA drivers,
+ index, the id string, short and long descriptions.
+
+version
+ Shows the version string and compile date.
+
+modules
+ Lists the module of each card
+
+devices
+ Lists the ALSA native device mappings.
+
+meminfo
+ Shows the status of allocated pages via ALSA drivers.
+ Appears only when CONFIG_SND_DEBUG=y.
+
+hwdep
+ Lists the currently available hwdep devices in format of
+ <card>-<device>: <name>
+
+pcm
+ Lists the currently available PCM devices in format of
+ <card>-<device>: <id>: <name> : <sub-streams>
+
+timer
+ Lists the currently available timer devices
+
+
+oss/devices
+ Lists the OSS device mappings.
+
+oss/sndstat
+ Provides the output compatible with /dev/sndstat.
+ You can symlink this to /dev/sndstat.
+
+
+Card Specific Files
+-------------------
+
+The card-specific files are found in /proc/asound/card* directories.
+Some drivers (e.g. cmipci) have their own proc entries for the
+register dump, etc (e.g. /proc/asound/card*/cmipci shows the register
+dump). These files would be really helpful for debugging.
+
+When PCM devices are available on this card, you can see directories
+like pcm0p or pcm1c. They hold the PCM information for each PCM
+stream. The number after 'pcm' is the PCM device number from 0, and
+the last 'p' or 'c' means playback or capture direction. The files in
+this subtree is described later.
+
+The status of MIDI I/O is found in midi* files. It shows the device
+name and the received/transmitted bytes through the MIDI device.
+
+When the card is equipped with AC97 codecs, there are codec97#*
+subdirectories (described later).
+
+When the OSS mixer emulation is enabled (and the module is loaded),
+oss_mixer file appears here, too. This shows the current mapping of
+OSS mixer elements to the ALSA control elements. You can change the
+mapping by writing to this device. Read OSS-Emulation.txt for
+details.
+
+
+PCM Proc Files
+--------------
+
+card*/pcm*/info
+ The general information of this PCM device: card #, device #,
+ substreams, etc.
+
+card*/pcm*/xrun_debug
+ This file appears when CONFIG_SND_DEBUG=y and
+ CONFIG_PCM_XRUN_DEBUG=y.
+ This shows the status of xrun (= buffer overrun/xrun) and
+ invalid PCM position debug/check of ALSA PCM middle layer.
+ It takes an integer value, can be changed by writing to this
+ file, such as
+
+ # echo 5 > /proc/asound/card0/pcm0p/xrun_debug
+
+ The value consists of the following bit flags:
+ bit 0 = Enable XRUN/jiffies debug messages
+ bit 1 = Show stack trace at XRUN / jiffies check
+ bit 2 = Enable additional jiffies check
+
+ When the bit 0 is set, the driver will show the messages to
+ kernel log when an xrun is detected. The debug message is
+ shown also when the invalid H/W pointer is detected at the
+ update of periods (usually called from the interrupt
+ handler).
+
+ When the bit 1 is set, the driver will show the stack trace
+ additionally. This may help the debugging.
+
+ Since 2.6.30, this option can enable the hwptr check using
+ jiffies. This detects spontaneous invalid pointer callback
+ values, but can be lead to too much corrections for a (mostly
+ buggy) hardware that doesn't give smooth pointer updates.
+ This feature is enabled via the bit 2.
+
+card*/pcm*/sub*/info
+ The general information of this PCM sub-stream.
+
+card*/pcm*/sub*/status
+ The current status of this PCM sub-stream, elapsed time,
+ H/W position, etc.
+
+card*/pcm*/sub*/hw_params
+ The hardware parameters set for this sub-stream.
+
+card*/pcm*/sub*/sw_params
+ The soft parameters set for this sub-stream.
+
+card*/pcm*/sub*/prealloc
+ The buffer pre-allocation information.
+
+card*/pcm*/sub*/xrun_injection
+ Triggers an XRUN to the running stream when any value is
+ written to this proc file. Used for fault injection.
+ This entry is write-only.
+
+AC97 Codec Information
+----------------------
+
+card*/codec97#*/ac97#?-?
+ Shows the general information of this AC97 codec chip, such as
+ name, capabilities, set up.
+
+card*/codec97#0/ac97#?-?+regs
+ Shows the AC97 register dump. Useful for debugging.
+
+ When CONFIG_SND_DEBUG is enabled, you can write to this file for
+ changing an AC97 register directly. Pass two hex numbers.
+ For example,
+
+ # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs
+
+
+USB Audio Streams
+-----------------
+
+card*/stream*
+ Shows the assignment and the current status of each audio stream
+ of the given card. This information is very useful for debugging.
+
+
+HD-Audio Codecs
+---------------
+
+card*/codec#*
+ Shows the general codec information and the attribute of each
+ widget node.
+
+card*/eld#*
+ Available for HDMI or DisplayPort interfaces.
+ Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink,
+ and describes its audio capabilities and configurations.
+
+ Some ELD fields may be modified by doing `echo name hex_value > eld#*`.
+ Only do this if you are sure the HDMI sink provided value is wrong.
+ And if that makes your HDMI audio work, please report to us so that we
+ can fix it in future kernel releases.
+
+
+Sequencer Information
+---------------------
+
+seq/drivers
+ Lists the currently available ALSA sequencer drivers.
+
+seq/clients
+ Shows the list of currently available sequencer clients and
+ ports. The connection status and the running status are shown
+ in this file, too.
+
+seq/queues
+ Lists the currently allocated/running sequencer queues.
+
+seq/timer
+ Lists the currently allocated/running sequencer timers.
+
+seq/oss
+ Lists the OSS-compatible sequencer stuffs.
+
+
+Help For Debugging?
+-------------------
+
+When the problem is related with PCM, first try to turn on xrun_debug
+mode. This will give you the kernel messages when and where xrun
+happened.
+
+If it's really a bug, report it with the following information:
+
+ - the name of the driver/card, show in /proc/asound/cards
+ - the register dump, if available (e.g. card*/cmipci)
+
+when it's a PCM problem,
+
+ - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM
+ sub-stream directory
+
+when it's a mixer problem,
+
+ - AC97 proc files, codec97#*/* files
+
+for USB audio/midi,
+
+ - output of lsusb -v
+ - stream* files in card directory
+
+
+The ALSA bug-tracking system is found at:
+
+ https://bugtrack.alsa-project.org/alsa-bug/
diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
new file mode 100644
index 000000000..67b2ea1cc
--- /dev/null
+++ b/Documentation/sound/alsa/README.maya44
@@ -0,0 +1,163 @@
+NOTE: The following is the original document of Rainer's patch that the
+current maya44 code based on. Some contents might be obsoleted, but I
+keep here as reference -- tiwai
+
+----------------------------------------------------------------
+
+STATE OF DEVELOPMENT:
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS:
+
+the following files were added:
+
+pci/ice1724/maya44.c - Maya44 specific code
+pci/ice1724/maya44.h
+pci/ice1724/ice1724.patch
+pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs
+include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+wtm.h
+vt1720_mobo.h
+revo.h
+prodigy192.h
+pontis.h
+phase.h
+maya44.h
+juli.h
+aureon.h
+amp.h
+envy24ht.h
+se.h
+prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES:
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES:
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+hw:0,0 input - stereo, analog input 1+2
+hw:0,0 output - stereo, analog output 1+2
+hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS:
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM: (digital) output level for channel 1+2
+PCM 1: same for channel 3+4
+
+Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
+ Make sure this is not turned on while any other source is connected to input 1/2.
+ It might damage the source and/or the maya44 card.
+
+Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1: same for channel 3+4.
+
+Crossmix: cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output: switch for S/PDIF output.
+ This is not supported by the ESI windows driver.
+ S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors:
+
+ These switches allow a direct digital routing from the ADCs to the DACs.
+ Each switch determines where the digital input data to one of the DACs comes from.
+ They are not supported by the ESI windows driver.
+ For normal operation, they should all be set to "PCM out".
+
+H/W: Output source channel 1
+H/W 1: Output source channel 2
+H/W 2: Output source channel 3
+H/W 3: Output source channel 4
+
+H/W 4 ... H/W 9: unknown function, left in to enable testing.
+ Possibly some of these control S/PDIF output(s).
+ If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+ "PCM out" -> DAC output of the corresponding channel (default setting)
+ "Input 1"...
+ "Input 4" -> direct routing from ADC output of the selected input channel
+
+
+--------
+
+Feb 14, 2008
+Rainer Zimmermann
+mail@lightshed.de
+
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
new file mode 100644
index 000000000..f4b5988f4
--- /dev/null
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -0,0 +1,356 @@
+
+ Sound Blaster Live mixer / default DSP code
+ ===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) IEC958 (S/PDIF) raw PCM
+--------------------------
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time.
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+2) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be wrapped
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC - digital to analog converter
+ADC - analog to digital converter
+I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
+ (this standard is used for connecting standalone DAC and ADC converters)
+LFE - low frequency effects (subwoofer signal)
+AC97 - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
+ Each of the synthesizer voices can feed its output to these accumulators
+ and the DSP microcontroller can operate with the resulting sum.
+
+
+name='Wave Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Wave Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+name='Wave Center Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='Wave LFE Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='Wave Capture Volume',index=0
+name='Wave Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Synth Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Synth Capture Volume',index=0
+name='Synth Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+name='Surround Capture Volume',index=0
+name='Surround Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Center Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='LFE Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='AC97 Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+********************************************************************************
+*** Note: This control should be zero for the standard operations, otherwise ***
+*** a digital loopback is activated. ***
+********************************************************************************
+
+name='AC97 Capture Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+********************************************************************************
+*** Note: This control should be 100 (maximal value), otherwise no analog ***
+*** inputs of the AC97 codec can be captured (recorded). ***
+********************************************************************************
+
+name='IEC958 TTL Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='IEC958 TTL Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='Zoom Video Playback Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Zoom Video Capture Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 LiveDrive Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 LiveDrive Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='IEC958 Coaxial Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 Coaxial Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line LiveDrive Playback Volume',index=0
+name='Line LiveDrive Playback Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+name='Line LiveDrive Capture Volume',index=1
+name='Line LiveDrive Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+name='Headphone Playback Volume',index=1
+
+This control attenuates the samples for the headphone output.
+
+name='Headphone Center Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+name='Headphone LFE Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+3) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+ 0 - mono, default 0xffff (no attenuation)
+ 1 - left, default 0xffff (no attenuation)
+ 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+ 0 - mono, A destination (FX-bus 0-15), default 0
+ 1 - mono, B destination (FX-bus 0-15), default 1
+ 2 - mono, C destination (FX-bus 0-15), default 2
+ 3 - mono, D destination (FX-bus 0-15), default 3
+ 4 - left, A destination (FX-bus 0-15), default 0
+ 5 - left, B destination (FX-bus 0-15), default 1
+ 6 - left, C destination (FX-bus 0-15), default 2
+ 7 - left, D destination (FX-bus 0-15), default 3
+ 8 - right, A destination (FX-bus 0-15), default 0
+ 9 - right, B destination (FX-bus 0-15), default 1
+ 10 - right, C destination (FX-bus 0-15), default 2
+ 11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
+more than once (it means 0=0 && 1=0 is an invalid combination).
+
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+ 0 - mono, A destination attn, default 255 (no attenuation)
+ 1 - mono, B destination attn, default 255 (no attenuation)
+ 2 - mono, C destination attn, default 0 (mute)
+ 3 - mono, D destination attn, default 0 (mute)
+ 4 - left, A destination attn, default 255 (no attenuation)
+ 5 - left, B destination attn, default 0 (mute)
+ 6 - left, C destination attn, default 0 (mute)
+ 7 - left, D destination attn, default 0 (mute)
+ 8 - right, A destination attn, default 0 (mute)
+ 9 - right, B destination attn, default 255 (no attenuation)
+ 10 - right, C destination attn, default 0 (mute)
+ 11 - right, D destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+ Files:
+ LM4545.pdf AC97 Codec
+
+ m2049.pdf The EMU10K1 Digital Audio Processor
+
+ hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+ Patent numbers:
+ WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+ streams
+
+ WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+ WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+ Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+ US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+ US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+ with a multiport memory onto which multiple asynchronous
+ digital sound samples can be concurrently loaded
+
+ US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+ US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+ US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000)
+ system bus with prioritization and modification of bus transfers
+ in accordance with loop ends and minimum block sizes
+
+ US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
+ pool of short term memory registers
+
+ US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
+ a common interrupt by associating programs to GP registers,
+ defining interrupt register, polling GP registers, and invoking
+ callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt
new file mode 100644
index 000000000..1b0ac06ba
--- /dev/null
+++ b/Documentation/sound/alsa/VIA82xx-mixer.txt
@@ -0,0 +1,8 @@
+
+ VIA82xx mixer
+ =============
+
+On many VIA82xx boards, the 'Input Source Select' mixer control does not work.
+Setting it to 'Input2' on such boards will cause recording to hang, or fail
+with EIO (input/output error) via OSS emulation. This control should be left
+at 'Input1' for such cards.
diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt
new file mode 100644
index 000000000..0fa40679b
--- /dev/null
+++ b/Documentation/sound/alsa/alsa-parameters.txt
@@ -0,0 +1,135 @@
+ ALSA Kernel Parameters
+ ~~~~~~~~~~~~~~~~~~~~~~
+
+See Documentation/kernel-parameters.txt for general information on
+specifying module parameters.
+
+This document may not be entirely up to date and comprehensive. The command
+"modinfo -p ${modulename}" shows a current list of all parameters of a loadable
+module. Loadable modules, after being loaded into the running kernel, also
+reveal their parameters in /sys/module/${modulename}/parameters/. Some of these
+parameters may be changed at runtime by the command
+"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}".
+
+
+ snd-ad1816a= [HW,ALSA]
+
+ snd-ad1848= [HW,ALSA]
+
+ snd-ali5451= [HW,ALSA]
+
+ snd-als100= [HW,ALSA]
+
+ snd-als4000= [HW,ALSA]
+
+ snd-azt2320= [HW,ALSA]
+
+ snd-cmi8330= [HW,ALSA]
+
+ snd-cmipci= [HW,ALSA]
+
+ snd-cs4231= [HW,ALSA]
+
+ snd-cs4232= [HW,ALSA]
+
+ snd-cs4236= [HW,ALSA]
+
+ snd-cs4281= [HW,ALSA]
+
+ snd-cs46xx= [HW,ALSA]
+
+ snd-dt019x= [HW,ALSA]
+
+ snd-dummy= [HW,ALSA]
+
+ snd-emu10k1= [HW,ALSA]
+
+ snd-ens1370= [HW,ALSA]
+
+ snd-ens1371= [HW,ALSA]
+
+ snd-es968= [HW,ALSA]
+
+ snd-es1688= [HW,ALSA]
+
+ snd-es18xx= [HW,ALSA]
+
+ snd-es1938= [HW,ALSA]
+
+ snd-es1968= [HW,ALSA]
+
+ snd-fm801= [HW,ALSA]
+
+ snd-gusclassic= [HW,ALSA]
+
+ snd-gusextreme= [HW,ALSA]
+
+ snd-gusmax= [HW,ALSA]
+
+ snd-hdsp= [HW,ALSA]
+
+ snd-ice1712= [HW,ALSA]
+
+ snd-intel8x0= [HW,ALSA]
+
+ snd-interwave= [HW,ALSA]
+
+ snd-interwave-stb=
+ [HW,ALSA]
+
+ snd-korg1212= [HW,ALSA]
+
+ snd-maestro3= [HW,ALSA]
+
+ snd-mpu401= [HW,ALSA]
+
+ snd-mtpav= [HW,ALSA]
+
+ snd-nm256= [HW,ALSA]
+
+ snd-opl3sa2= [HW,ALSA]
+
+ snd-opti92x-ad1848=
+ [HW,ALSA]
+
+ snd-opti92x-cs4231=
+ [HW,ALSA]
+
+ snd-opti93x= [HW,ALSA]
+
+ snd-pmac= [HW,ALSA]
+
+ snd-rme32= [HW,ALSA]
+
+ snd-rme96= [HW,ALSA]
+
+ snd-rme9652= [HW,ALSA]
+
+ snd-sb8= [HW,ALSA]
+
+ snd-sb16= [HW,ALSA]
+
+ snd-sbawe= [HW,ALSA]
+
+ snd-serial= [HW,ALSA]
+
+ snd-sgalaxy= [HW,ALSA]
+
+ snd-sonicvibes= [HW,ALSA]
+
+ snd-sun-amd7930=
+ [HW,ALSA]
+
+ snd-sun-cs4231= [HW,ALSA]
+
+ snd-trident= [HW,ALSA]
+
+ snd-usb-audio= [HW,ALSA,USB]
+
+ snd-via82xx= [HW,ALSA]
+
+ snd-virmidi= [HW,ALSA]
+
+ snd-wavefront= [HW,ALSA]
+
+ snd-ymfpci= [HW,ALSA]
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
new file mode 100644
index 000000000..630c492c3
--- /dev/null
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -0,0 +1,234 @@
+ compress_offload.txt
+ =====================
+ Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
+ Vinod Koul <vinod.koul@linux.intel.com>
+
+Overview
+
+Since its early days, the ALSA API was defined with PCM support or
+constant bitrates payloads such as IEC61937 in mind. Arguments and
+returned values in frames are the norm, making it a challenge to
+extend the existing API to compressed data streams.
+
+In recent years, audio digital signal processors (DSP) were integrated
+in system-on-chip designs, and DSPs are also integrated in audio
+codecs. Processing compressed data on such DSPs results in a dramatic
+reduction of power consumption compared to host-based
+processing. Support for such hardware has not been very good in Linux,
+mostly because of a lack of a generic API available in the mainline
+kernel.
+
+Rather than requiring a compatibility break with an API change of the
+ALSA PCM interface, a new 'Compressed Data' API is introduced to
+provide a control and data-streaming interface for audio DSPs.
+
+The design of this API was inspired by the 2-year experience with the
+Intel Moorestown SOC, with many corrections required to upstream the
+API in the mainline kernel instead of the staging tree and make it
+usable by others.
+
+Requirements
+
+The main requirements are:
+
+- separation between byte counts and time. Compressed formats may have
+ a header per file, per frame, or no header at all. The payload size
+ may vary from frame-to-frame. As a result, it is not possible to
+ estimate reliably the duration of audio buffers when handling
+ compressed data. Dedicated mechanisms are required to allow for
+ reliable audio-video synchronization, which requires precise
+ reporting of the number of samples rendered at any given time.
+
+- Handling of multiple formats. PCM data only requires a specification
+ of the sampling rate, number of channels and bits per sample. In
+ contrast, compressed data comes in a variety of formats. Audio DSPs
+ may also provide support for a limited number of audio encoders and
+ decoders embedded in firmware, or may support more choices through
+ dynamic download of libraries.
+
+- Focus on main formats. This API provides support for the most
+ popular formats used for audio and video capture and playback. It is
+ likely that as audio compression technology advances, new formats
+ will be added.
+
+- Handling of multiple configurations. Even for a given format like
+ AAC, some implementations may support AAC multichannel but HE-AAC
+ stereo. Likewise WMA10 level M3 may require too much memory and cpu
+ cycles. The new API needs to provide a generic way of listing these
+ formats.
+
+- Rendering/Grabbing only. This API does not provide any means of
+ hardware acceleration, where PCM samples are provided back to
+ user-space for additional processing. This API focuses instead on
+ streaming compressed data to a DSP, with the assumption that the
+ decoded samples are routed to a physical output or logical back-end.
+
+ - Complexity hiding. Existing user-space multimedia frameworks all
+ have existing enums/structures for each compressed format. This new
+ API assumes the existence of a platform-specific compatibility layer
+ to expose, translate and make use of the capabilities of the audio
+ DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
+ applications are not supposed to make use of this API.
+
+
+Design
+
+The new API shares a number of concepts with the PCM API for flow
+control. Start, pause, resume, drain and stop commands have the same
+semantics no matter what the content is.
+
+The concept of memory ring buffer divided in a set of fragments is
+borrowed from the ALSA PCM API. However, only sizes in bytes can be
+specified.
+
+Seeks/trick modes are assumed to be handled by the host.
+
+The notion of rewinds/forwards is not supported. Data committed to the
+ring buffer cannot be invalidated, except when dropping all buffers.
+
+The Compressed Data API does not make any assumptions on how the data
+is transmitted to the audio DSP. DMA transfers from main memory to an
+embedded audio cluster or to a SPI interface for external DSPs are
+possible. As in the ALSA PCM case, a core set of routines is exposed;
+each driver implementer will have to write support for a set of
+mandatory routines and possibly make use of optional ones.
+
+The main additions are
+
+- get_caps
+This routine returns the list of audio formats supported. Querying the
+codecs on a capture stream will return encoders, decoders will be
+listed for playback streams.
+
+- get_codec_caps For each codec, this routine returns a list of
+capabilities. The intent is to make sure all the capabilities
+correspond to valid settings, and to minimize the risks of
+configuration failures. For example, for a complex codec such as AAC,
+the number of channels supported may depend on a specific profile. If
+the capabilities were exposed with a single descriptor, it may happen
+that a specific combination of profiles/channels/formats may not be
+supported. Likewise, embedded DSPs have limited memory and cpu cycles,
+it is likely that some implementations make the list of capabilities
+dynamic and dependent on existing workloads. In addition to codec
+settings, this routine returns the minimum buffer size handled by the
+implementation. This information can be a function of the DMA buffer
+sizes, the number of bytes required to synchronize, etc, and can be
+used by userspace to define how much needs to be written in the ring
+buffer before playback can start.
+
+- set_params
+This routine sets the configuration chosen for a specific codec. The
+most important field in the parameters is the codec type; in most
+cases decoders will ignore other fields, while encoders will strictly
+comply to the settings
+
+- get_params
+This routines returns the actual settings used by the DSP. Changes to
+the settings should remain the exception.
+
+- get_timestamp
+The timestamp becomes a multiple field structure. It lists the number
+of bytes transferred, the number of samples processed and the number
+of samples rendered/grabbed. All these values can be used to determine
+the average bitrate, figure out if the ring buffer needs to be
+refilled or the delay due to decoding/encoding/io on the DSP.
+
+Note that the list of codecs/profiles/modes was derived from the
+OpenMAX AL specification instead of reinventing the wheel.
+Modifications include:
+- Addition of FLAC and IEC formats
+- Merge of encoder/decoder capabilities
+- Profiles/modes listed as bitmasks to make descriptors more compact
+- Addition of set_params for decoders (missing in OpenMAX AL)
+- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
+- Addition of format information for WMA
+- Addition of encoding options when required (derived from OpenMAX IL)
+- Addition of rateControlSupported (missing in OpenMAX AL)
+
+Gapless Playback
+================
+When playing thru an album, the decoders have the ability to skip the encoder
+delay and padding and directly move from one track content to another. The end
+user can perceive this as gapless playback as we dont have silence while
+switching from one track to another
+
+Also, there might be low-intensity noises due to encoding. Perfect gapless is
+difficult to reach with all types of compressed data, but works fine with most
+music content. The decoder needs to know the encoder delay and encoder padding.
+So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
+and are not present by default in the bitstream, hence the need for a new
+interface to pass this information to the DSP. Also DSP and userspace needs to
+switch from one track to another and start using data for second track.
+
+The main additions are:
+
+- set_metadata
+This routine sets the encoder delay and encoder padding. This can be used by
+decoder to strip the silence. This needs to be set before the data in the track
+is written.
+
+- set_next_track
+This routine tells DSP that metadata and write operation sent after this would
+correspond to subsequent track
+
+- partial drain
+This is called when end of file is reached. The userspace can inform DSP that
+EOF is reached and now DSP can start skipping padding delay. Also next write
+data would belong to next track
+
+Sequence flow for gapless would be:
+- Open
+- Get caps / codec caps
+- Set params
+- Set metadata of the first track
+- Fill data of the first track
+- Trigger start
+- User-space finished sending all,
+- Indicaite next track data by sending set_next_track
+- Set metadata of the next track
+- then call partial_drain to flush most of buffer in DSP
+- Fill data of the next track
+- DSP switches to second track
+(note: order for partial_drain and write for next track can be reversed as well)
+
+Not supported:
+
+- Support for VoIP/circuit-switched calls is not the target of this
+ API. Support for dynamic bit-rate changes would require a tight
+ coupling between the DSP and the host stack, limiting power savings.
+
+- Packet-loss concealment is not supported. This would require an
+ additional interface to let the decoder synthesize data when frames
+ are lost during transmission. This may be added in the future.
+
+- Volume control/routing is not handled by this API. Devices exposing a
+ compressed data interface will be considered as regular ALSA devices;
+ volume changes and routing information will be provided with regular
+ ALSA kcontrols.
+
+- Embedded audio effects. Such effects should be enabled in the same
+ manner, no matter if the input was PCM or compressed.
+
+- multichannel IEC encoding. Unclear if this is required.
+
+- Encoding/decoding acceleration is not supported as mentioned
+ above. It is possible to route the output of a decoder to a capture
+ stream, or even implement transcoding capabilities. This routing
+ would be enabled with ALSA kcontrols.
+
+- Audio policy/resource management. This API does not provide any
+ hooks to query the utilization of the audio DSP, nor any preemption
+ mechanisms.
+
+- No notion of underrun/overrun. Since the bytes written are compressed
+ in nature and data written/read doesn't translate directly to
+ rendered output in time, this does not deal with underrun/overrun and
+ maybe dealt in user-library
+
+Credits:
+- Mark Brown and Liam Girdwood for discussions on the need for this API
+- Harsha Priya for her work on intel_sst compressed API
+- Rakesh Ughreja for valuable feedback
+- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
+ demonstrating and quantifying the benefits of audio offload on a
+ real platform.
diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt
new file mode 100644
index 000000000..751d45036
--- /dev/null
+++ b/Documentation/sound/alsa/emu10k1-jack.txt
@@ -0,0 +1,74 @@
+This document is a guide to using the emu10k1 based devices with JACK for low
+latency, multichannel recording functionality. All of my recent work to allow
+Linux users to use the full capabilities of their hardware has been inspired
+by the kX Project. Without their work I never would have discovered the true
+power of this hardware.
+
+ http://www.kxproject.com
+ - Lee Revell, 2005.03.30
+
+Low latency, multichannel audio with JACK and the emu10k1/emu10k2
+-----------------------------------------------------------------
+
+Until recently, emu10k1 users on Linux did not have access to the same low
+latency, multichannel features offered by the "kX ASIO" feature of their
+Windows driver. As of ALSA 1.0.9 this is no more!
+
+For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
+channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
+even 32 (0.66ms) frames should work well.
+
+The configuration is slightly more involved than on Windows, as you have to
+select the correct device for JACK to use. Actually, for qjackctl users it's
+fairly self explanatory - select Duplex, then for capture and playback select
+the multichannel devices, set the in and out channels to 16, and the sample
+rate to 48000Hz. The command line looks like this:
+
+/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
+
+This will give you 16 input ports and 16 output ports.
+
+The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
+Audigy). The mapping from FX bus to physical output is described in
+SB-Live-mixer.txt (or Audigy-mixer.txt).
+
+The 16 input ports are connected to the 16 physical inputs. Contrary to
+popular belief, all emu10k1 cards are multichannel cards. Which of these
+input channels have physical inputs connected to them depends on the card
+model. Trial and error is highly recommended; the pinout diagrams
+for the card have been reverse engineered by some enterprising kX users and are
+available on the internet. Meterbridge is helpful here, and the kX forums are
+packed with useful information.
+
+Each input port will either correspond to a digital (SPDIF) input, an analog
+input, or nothing. The one exception is the SBLive! 5.1. On these devices,
+the second and third input ports are wired to the center/LFE output. You will
+still see 16 capture channels, but only 14 are available for recording inputs.
+
+This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
+ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
+channels.
+
+/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
+--------------------------------------------
+JACK Epilog FXBUS2(nr)
+--------------------------------------------
+capture_1 asio14 FXBUS2(0xe)
+capture_2 asio15 FXBUS2(0xf)
+capture_3 asio0 FXBUS2(0x0)
+~capture_4 Center EXTOUT(0x11) // mapped to by Center
+~capture_5 LFE EXTOUT(0x12) // mapped to by LFE
+capture_6 asio3 FXBUS2(0x3)
+capture_7 asio4 FXBUS2(0x4)
+capture_8 asio5 FXBUS2(0x5)
+capture_9 asio6 FXBUS2(0x6)
+capture_10 asio7 FXBUS2(0x7)
+capture_11 asio8 FXBUS2(0x8)
+capture_12 asio9 FXBUS2(0x9)
+capture_13 asio10 FXBUS2(0xa)
+capture_14 asio11 FXBUS2(0xb)
+capture_15 asio12 FXBUS2(0xc)
+capture_16 asio13 FXBUS2(0xd)
+*/
+
+TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
new file mode 100644
index 000000000..de8efbc7e
--- /dev/null
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -0,0 +1,322 @@
+Notes on Universal Interface for Intel High Definition Audio Codec
+------------------------------------------------------------------
+
+Takashi Iwai <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
new file mode 100644
index 000000000..7ba31948d
--- /dev/null
+++ b/Documentation/sound/alsa/hdspm.txt
@@ -0,0 +1,362 @@
+Software Interface ALSA-DSP MADI Driver
+
+(translated from German, so no good English ;-),
+2004 - winfried ritsch
+
+
+
+ Full functionality has been added to the driver. Since some of
+ the Controls and startup-options are ALSA-Standard and only the
+ special Controls are described and discussed below.
+
+
+ hardware functionality:
+
+
+ Audio transmission:
+
+ number of channels -- depends on transmission mode
+
+ The number of channels chosen is from 1..Nmax. The reason to
+ use for a lower number of channels is only resource allocation,
+ since unused DMA channels are disabled and less memory is
+ allocated. So also the throughput of the PCI system can be
+ scaled. (Only important for low performance boards).
+
+ Single Speed -- 1..64 channels
+
+ (Note: Choosing the 56channel mode for transmission or as
+ receiver, only 56 are transmitted/received over the MADI, but
+ all 64 channels are available for the mixer, so channel count
+ for the driver)
+
+ Double Speed -- 1..32 channels
+
+ Note: Choosing the 56-channel mode for
+ transmission/receive-mode , only 28 are transmitted/received
+ over the MADI, but all 32 channels are available for the mixer,
+ so channel count for the driver
+
+
+ Quad Speed -- 1..16 channels
+
+ Note: Choosing the 56-channel mode for
+ transmission/receive-mode , only 14 are transmitted/received
+ over the MADI, but all 16 channels are available for the mixer,
+ so channel count for the driver
+
+ Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+ Sample Rates --
+
+ Single Speed -- 32000, 44100, 48000
+
+ Double Speed -- 64000, 88200, 96000 (untested)
+
+ Quad Speed -- 128000, 176400, 192000 (untested)
+
+ access-mode -- MMAP (memory mapped), Not interleaved
+ (PCM_NON-INTERLEAVED)
+
+ buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+ fragments -- 2
+
+ Hardware-pointer -- 2 Modi
+
+
+ The Card supports the readout of the actual Buffer-pointer,
+ where DMA reads/writes. Since of the bulk mode of PCI it is only
+ 64 Byte accurate. SO it is not really usable for the
+ ALSA-mid-level functions (here the buffer-ID gives a better
+ result), but if MMAP is used by the application. Therefore it
+ can be configured at load-time with the parameter
+ precise-pointer.
+
+
+ (Hint: Experimenting I found that the pointer is maximum 64 to
+ large never to small. So if you subtract 64 you always have a
+ safe pointer for writing, which is used on this mode inside
+ ALSA. In theory now you can get now a latency as low as 16
+ Samples, which is a quarter of the interrupt possibilities.)
+
+ Precise Pointer -- off
+ interrupt used for pointer-calculation
+
+ Precise Pointer -- on
+ hardware pointer used.
+
+ Controller:
+
+
+ Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+ use the standard mixer-controls, since this would break most of
+ (especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+ provided by a 2-dimensional controller using the
+ hwdep-interface.
+
+ Also all 128+256 Peak and RMS-Meter can be accessed via the
+ hwdep-interface. Since it could be a performance problem always
+ copying and converting Peak and RMS-Levels even if you just need
+ one, I decided to export the hardware structure, so that of
+ needed some driver-guru can implement a memory-mapping of mixer
+ or peak-meters over ioctl, or also to do only copying and no
+ conversion. A test-application shows the usage of the controller.
+
+ Latency Controls --- not implemented !!!
+
+
+ Note: Within the windows-driver the latency is accessible of a
+ control-panel, but buffer-sizes are controlled with ALSA from
+ hwparams-calls and should not be changed in run-state, I did not
+ implement it here.
+
+
+ System Clock -- suspended !!!!
+
+ Name -- "System Clock Mode"
+
+ Access -- Read Write
+
+ Values -- "Master" "Slave"
+
+
+ !!!! This is a hardware-function but is in conflict with the
+ Clock-source controller, which is a kind of ALSA-standard. I
+ makes sense to set the card to a special mode (master at some
+ frequency or slave), since even not using an Audio-application
+ a studio should have working synchronisations setup. So use
+ Clock-source-controller instead !!!!
+
+ Clock Source
+
+ Name -- "Sample Clock Source"
+
+ Access -- Read Write
+
+ Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+ "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+ "Internal 96.0 kHz"
+
+ Choose between Master at a specific Frequency and so also the
+ Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref"
+
+
+ !!!! This is no pure hardware function but was implemented by
+ ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+ Preferred Sync Ref
+
+ Name -- "Preferred Sync Reference"
+
+ Access -- Read Write
+
+ Values -- "Word" "MADI"
+
+
+ Within the Auto-sync-Mode the preferred Sync Source can be
+ chosen. If it is not available another is used if possible.
+
+ Note: Since MADI has a much higher bit-rate than word-clock, the
+ card should synchronise better in MADI Mode. But since the
+ RME-PLL is very good, there are almost no problems with
+ word-clock too. I never found a difference.
+
+
+ TX 64 channel ---
+
+ Name -- "TX 64 channels mode"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ Using 64-channel-modus (1) or 56-channel-modus for
+ MADI-transmission (0).
+
+
+ Note: This control is for output only. Input-mode is detected
+ automatically from hardware sending MADI.
+
+
+ Clear TMS ---
+
+ Name -- "Clear Track Marker"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+
+ Don't use to lower 5 Audio-bits on AES as additional Bits.
+
+
+ Safe Mode oder Auto Input ---
+
+ Name -- "Safe Mode"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ (default on)
+
+ If on (1), then if either the optical or coaxial connection
+ has a failure, there is a takeover to the working one, with no
+ sample failure. Its only useful if you use the second as a
+ backup connection.
+
+ Input ---
+
+ Name -- "Input Select"
+
+ Access -- Read Write
+
+ Values -- optical coaxial
+
+
+ Choosing the Input, optical or coaxial. If Safe-mode is active,
+ this is the preferred Input.
+
+-------------- Mixer ----------------------
+
+ Mixer
+
+ Name -- "Mixer"
+
+ Access -- Read Write
+
+ Values - <channel-number 0-127> <Value 0-65535>
+
+
+ Here as a first value the channel-index is taken to get/set the
+ corresponding mixer channel, where 0-63 are the input to output
+ fader and 64-127 the playback to outputs fader. Value 0
+ is channel muted 0 and 32768 an amplification of 1.
+
+ Chn 1-64
+
+ fast mixer for the ALSA-mixer utils. The diagonal of the
+ mixer-matrix is implemented from playback to output.
+
+
+ Line Out
+
+ Name -- "Line Out"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ Switching on and off the analog out, which has nothing to do
+ with mixing or routing. the analog outs reflects channel 63,64.
+
+
+--- information (only read access):
+
+ Sample Rate
+
+ Name -- "System Sample Rate"
+
+ Access -- Read-only
+
+ getting the sample rate.
+
+
+ External Rate measured
+
+ Name -- "External Rate"
+
+ Access -- Read only
+
+
+ Should be "Autosync Rate", but Name used is
+ ALSA-Scheme. External Sample frequency liked used on Autosync is
+ reported.
+
+
+ MADI Sync Status
+
+ Name -- "MADI Sync Lock Status"
+
+ Access -- Read
+
+ Values -- 0,1,2
+
+ MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+ Word Clock Sync Status
+
+ Name -- "Word Clock Lock Status"
+
+ Access -- Read
+
+ Values -- 0,1,2
+
+ Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+ AutoSync
+
+ Name -- "AutoSync Reference"
+
+ Access -- Read
+
+ Values -- "WordClock", "MADI", "None"
+
+ Sync-Reference is either "WordClock", "MADI" or none.
+
+ RX 64ch --- noch nicht implementiert
+
+ MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+ AB_inp --- not tested
+
+ Used input for Auto-Input.
+
+
+ actual Buffer Position --- not implemented
+
+ !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter:
+
+ index int array (min = 1, max = 8),
+ "Index value for RME HDSPM interface." card-index within ALSA
+
+ note: ALSA-standard
+
+ id string array (min = 1, max = 8),
+ "ID string for RME HDSPM interface."
+
+ note: ALSA-standard
+
+ enable int array (min = 1, max = 8),
+ "Enable/disable specific HDSPM sound-cards."
+
+ note: ALSA-standard
+
+ precise_ptr int array (min = 1, max = 8),
+ "Enable precise pointer, or disable."
+
+ note: Use only when the application supports this (which is a special case).
+
+ line_outs_monitor int array (min = 1, max = 8),
+ "Send playback streams to analog outs by default."
+
+
+ note: each playback channel is mixed to the same numbered output
+ channel (routed). This is against the ALSA-convention, where all
+ channels have to be muted on after loading the driver, but was
+ used before on other cards, so i historically use it again)
+
+
+
+ enable_monitor int array (min = 1, max = 8),
+ "Enable Analog Out on Channel 63/64 by default."
+
+ note: here the analog output is enabled (but not routed).
diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt
new file mode 100644
index 000000000..9657e8099
--- /dev/null
+++ b/Documentation/sound/alsa/powersave.txt
@@ -0,0 +1,41 @@
+Notes on Power-Saving Mode
+==========================
+
+AC97 and HD-audio drivers have the automatic power-saving mode.
+This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE
+and CONFIG_SND_HDA_POWER_SAVE options, respectively.
+
+With the automatic power-saving, the driver turns off the codec power
+appropriately when no operation is required. When no applications use
+the device and/or no analog loopback is set, the power disablement is
+done fully or partially. It'll save a certain power consumption, thus
+good for laptops (even for desktops).
+
+The time-out for automatic power-off can be specified via power_save
+module option of snd-ac97-codec and snd-hda-intel modules. Specify
+the time-out value in seconds. 0 means to disable the automatic
+power-saving. The default value of timeout is given via
+CONFIG_SND_AC97_POWER_SAVE_DEFAULT and
+CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1
+(the minimum value) isn't recommended because many applications try to
+reopen the device frequently. 10 would be a good choice for normal
+operations.
+
+The power_save option is exported as writable. This means you can
+adjust the value via sysfs on the fly. For example, to turn on the
+automatic power-save mode with 10 seconds, write to
+/sys/modules/snd_ac97_codec/parameters/power_save (usually as root):
+
+ # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save
+
+
+Note that you might hear click noise/pop when changing the power
+state. Also, it often takes certain time to wake up from the
+power-down to the active state. These are often hardly to fix, so
+don't report extra bug reports unless you have a fix patch ;-)
+
+For HD-audio interface, there is another module option,
+power_save_controller. This enables/disables the power-save mode of
+the controller side. Setting this on may reduce a bit more power
+consumption, but might result in longer wake-up time and click noise.
+Try to turn it off when you experience such a thing too often.
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
new file mode 100644
index 000000000..9663b45f6
--- /dev/null
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -0,0 +1,409 @@
+<!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">&lt;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>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; OSS sequencer emulation version 0.1.8
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA client number 63
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA receiver port 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of applications: 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of synth devices: 1
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synth 0: [EMU8000]
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; type 0x1 : subtype 0x20 : voices 32
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capabilties : ioctl enabled / load_patch enabled
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of MIDI devices: 3
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 0: [Emu8000 Port-0] ALSA port 65:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 1: [Emu8000 Port-1] ALSA port 65:1
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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,
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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:
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*open)(snd_seq_oss_arg_t *p, void *closure);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*close)(snd_seq_oss_arg_t *p);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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 {
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int app_index;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int file_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int seq_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_addr_t addr;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *private_data;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int event_passing;
+};</PRE>
+The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and
+<TT>seq_mode</TT>
+are initialized by OSS sequencer. The <TT>app_index</TT> is the application
+index which is unique to each application opening OSS sequencer. The
+<TT>file_mode</TT>
+is bit-flags indicating the file operation mode. See
+<TT>seq_oss.h</TT>
+for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In
+the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used.
+<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be
+filled by the synth driver at open callback. The <TT>addr</TT> contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for <TT>private_data</TT>, it must be released
+in close callback by itself.
+<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on
+/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT>
+mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT>
+mode checks the note above 128 and regards it as key pressure event (mainly
+for Emu8000 driver).
+<H4>
+7.2.1. Open Callback</H4>
+The <TT>open</TT> is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+<OL>
+<LI>
+Allocate private data record.</LI>
+
+<LI>
+Create an ALSA sequencer port.</LI>
+
+<LI>
+Set the new port address on arg->addr.</LI>
+
+<LI>
+Set the private data record pointer on arg->private_data.</LI>
+</OL>
+Note that the type bit-flags in port_info of this synth port must NOT contain
+<TT>TYPE_MIDI_GENERIC</TT>
+bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT>
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+<H4>
+7.2.2 Ioctl Callback</H4>
+The <TT>ioctl</TT> callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+<UL>
+<LI>
+<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI>
+
+<BR>reset all samples on memory -- return 0
+<LI>
+<TT>IOCTL_SYNTH_MEMAVL</TT></LI>
+
+<BR>return the available memory size
+<LI>
+<TT>FM_4OP_ENABLE</TT></LI>
+
+<BR>can be ignored usually</UL>
+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+<H4>
+7.2.3 Load_Patch Callback</H4>
+The <TT>load_patch</TT> callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+<H4>
+7.2.4 Close Callback</H4>
+The <TT>close</TT> callback is called when this device is closed by the
+application. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+<H4>
+7.2.5 Reset Callback</H4>
+The <TT>reset</TT> callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the
+port.
+<H3>
+7.3 Events</H3>
+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be implemented
+by the driver:
+<BR>&nbsp;
+<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
new file mode 100644
index 000000000..c1919559d
--- /dev/null
+++ b/Documentation/sound/alsa/serial-u16550.txt
@@ -0,0 +1,88 @@
+
+ Serial UART 16450/16550 MIDI driver
+ ===================================
+
+The adaptor module parameter allows you to select either:
+
+ 0 - Roland Soundcanvas support (default)
+ 1 - Midiator MS-124T support (1)
+ 2 - Midiator MS-124W S/A mode (2)
+ 3 - MS-124W M/B mode support (3)
+ 4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1. Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors. Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+ speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16. The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports. M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs.
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported.
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with. The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow. The number of
+inputs is specified by the ins parameter.
diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
index 000000000..c9679264c
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DAI.txt
@@ -0,0 +1,56 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+ AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at :-
+http://www.intel.com/p/en_US/business/design
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
+ transition.
+
+ o Left Justified - MSB is transmitted on transition of LRC.
+
+ o Right Justified - MSB is transmitted sample size BCLKs before LRC
+ transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, which can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus simultaneously (this
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
new file mode 100644
index 000000000..0110180b7
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DPCM.txt
@@ -0,0 +1,380 @@
+Dynamic PCM
+===========
+
+1. Description
+==============
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the patch used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+
+| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
+
+ *************
+PCM0 <------------> * * <----DAI0-----> Codec Headset
+ * *
+PCM1 <------------> * * <----DAI1-----> Codec Speakers
+ * DSP *
+PCM2 <------------> * * <----DAI2-----> MODEM
+ * *
+PCM3 <------------> * * <----DAI3-----> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+
+ *************
+PCM0 <============> * * <====DAI0=====> Codec Headset
+ * *
+PCM1 <------------> * * <----DAI1-----> Codec Speakers
+ * DSP *
+PCM2 <------------> * * <----DAI2-----> MODEM
+ * *
+PCM3 <------------> * * <----DAI3-----> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+
+ *************
+PCM0 <============> * * <----DAI0-----> Codec Headset
+ * *
+PCM1 <------------> * * <====DAI1=====> Codec Speakers
+ * DSP *
+PCM2 <------------> * * <----DAI2-----> MODEM
+ * *
+PCM3 <------------> * * <----DAI3-----> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+The audio driver processes this as follows :-
+
+ 1) Machine driver receives Jack removal event.
+
+ 2) Machine driver OR audio HAL disables the Headset path.
+
+ 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+ for headset since the path is now disabled.
+
+ 4) Machine driver or audio HAL enables the speaker path.
+
+ 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+ trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+ 1) Define the FE and BE DAI links.
+
+ 2) Define any FE/BE PCM operations.
+
+ 3) Define widget graph connections.
+
+
+1 FE and BE DAI links
+---------------------
+
+| Front End PCMs | SoC DSP | Back End DAIs | Audio devices |
+
+ *************
+PCM0 <------------> * * <----DAI0-----> Codec Headset
+ * *
+PCM1 <------------> * * <----DAI1-----> Codec Speakers
+ * DSP *
+PCM2 <------------> * * <----DAI2-----> MODEM
+ * *
+PCM3 <------------> * * <----DAI3-----> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+ {
+ .name = "PCM0 System",
+ .stream_name = "System Playback",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "dsp-audio",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ .....< other FE and BE DAI links here >
+};
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream
+directions should also be set with the "dpcm_playback" and "dpcm_capture"
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+
+static struct snd_soc_dai_link machine_dais[] = {
+ .....< FE DAI links here >
+ {
+ .name = "Codec Headset",
+ .cpu_dai_name = "ssp-dai.0",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "rt5640.0-001c",
+ .codec_dai_name = "rt5640-aif1",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = hswult_ssp0_fixup,
+ .ops = &haswell_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ .....< other BE DAI links here >
+};
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the "no_pcm" flag to mark it has a BE and sets flags for supported stream
+directions using "dpcm_playback" and "dpcm_capture" above.
+
+The BE has also flags set for ignoring suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+
+ *************
+PCM0 <------------> * * <----DAI0-----> Codec Headset
+ * *
+PCM1 <------------> * * <----DAI1-----> Codec Speakers
+ * DSP *
+PCM2 <------------> * * <====DAI2=====> MODEM
+ * *
+PCM3 <------------> * * <====DAI3=====> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the code is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+2 FE/BE PCM operations
+----------------------
+
+The BE above also exports some PCM operations and a "fixup" callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+
+static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set DAI0 to 16 bit */
+ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+3 Widget graph connections
+--------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+
+/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */
+{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+ 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+ 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+ 3) DAPM widgets from DSP graph.
+
+ 4) Mixers for gains, routing, etc.
+
+ 5) DMA configuration.
+
+ 6) BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+
+SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+
+
+ *************
+PCM0 <------------> * * <----DAI0-----> Codec Headset
+ * *
+PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic
+ * DSP *
+PCM2 <------------> * * <====DAI2=====> MODEM
+ * *
+PCM3 <------------> * * <----DAI3-----> BT
+ * *
+ * * <----DAI4-----> DMIC
+ * *
+ * * <----DAI5-----> FM
+ *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link
+ is enabled or disabled by the state of the DAPM graph. This usually means
+ there is a mixer control that can be used to connect or disconnect the path
+ between both DAIs.
+
+ 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+ graph. Control is then carried out by the FE as regular PCM operations.
+ This method gives more control over the DAI links, but requires much more
+ userspace code to control the link. Its recommended to use CODEC<->CODEC
+ unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+
+static const struct snd_soc_pcm_stream dai_params = {
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static struct snd_soc_dai_link dais[] = {
+ < ... more DAI links above ... >
+ {
+ .name = "MODEM",
+ .stream_name = "MODEM",
+ .cpu_dai_name = "dai2",
+ .codec_dai_name = "modem-aif1",
+ .codec_name = "modem",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBM_CFM,
+ .params = &dai_params,
+ }
+ < ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
+
+
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
new file mode 100644
index 000000000..b1300162e
--- /dev/null
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -0,0 +1,51 @@
+Audio Clocking
+==============
+
+This text describes the audio clocking terms in ASoC and digital audio in
+general. Note: Audio clocking can be complex!
+
+
+Master Clock
+------------
+
+Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
+or SYSCLK). This audio master clock can be derived from a number of sources
+(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
+audio playback and capture sample rates.
+
+Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at a set frequency (i.e. crystals).
+
+
+DAI Clocks
+----------
+The Digital Audio Interface is usually driven by a Bit Clock (often referred to
+as BCLK). This clock is used to drive the digital audio data across the link
+between the codec and CPU.
+
+The DAI also has a frame clock to signal the start of each audio frame. This
+clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
+runs at exactly the sample rate (LRC = Rate).
+
+Bit Clock can be generated as follows:-
+
+BCLK = MCLK / x
+
+ or
+
+BCLK = LRC * x
+
+ or
+
+BCLK = LRC * Channels * Word Size
+
+This relationship depends on the codec or SoC CPU in particular. In general
+it is best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and word size) to save on power.
+
+It is also desirable to use the codec (if possible) to drive (or master) the
+audio clocks as it usually gives more accurate sample rates than the CPU.
+
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 000000000..db5f9c9ae
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,179 @@
+ASoC Codec Class Driver
+=======================
+
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
+
+Each codec class driver *must* provide the following features:-
+
+ 1) Codec DAI and PCM configuration
+ 2) Codec control IO - using RegMap API
+ 3) Mixers and audio controls
+ 4) Codec audio operations
+ 5) DAPM description.
+ 6) DAPM event handler.
+
+Optionally, codec drivers can also provide:-
+
+ 7) DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Codec DAI and PCM configuration
+-----------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+ .name = "wm8731-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .ops = &wm8731_dai_ops,
+ .symmetric_rates = 1,
+};
+
+
+2 - Codec control IO
+--------------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
+
+
+3 - Mixers and audio controls
+-----------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+
+ #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+
+ xname = Control name e.g. "Playback Volume"
+ reg = codec register
+ shift = control bit(s) offset in register
+ mask = control bit size(s) e.g. mask of 7 = 3 bits
+ invert = the control is inverted
+
+Other macros include:-
+
+ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+
+ #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+
+ #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+
+ xreg = register
+ xshift = control bit(s) offset in register
+ xmask = control bit(s) size
+ xtexts = pointer to array of strings that describe each setting
+
+ #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+4 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following ALSA PCM operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
+ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+ int (*hw_free)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+};
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+5 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+6 - DAPM event handler
+----------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+
+ SNDRV_CTL_POWER_D0: /* full On */
+ /* vref/mid, clk and osc on, active */
+
+ SNDRV_CTL_POWER_D1: /* partial On */
+ SNDRV_CTL_POWER_D2: /* partial On */
+
+ SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* everything off except vref/vmid, inactive */
+
+ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+7 - Codec DAC digital mute control
+----------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise. The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+ else
+ snd_soc_write(codec, WM8974_DAC, mute_reg);
+ return 0;
+}
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
new file mode 100644
index 000000000..6faab4880
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,305 @@
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+1. Description
+==============
+
+Dynamic Audio Power Management (DAPM) is designed to allow portable
+Linux devices to use the minimum amount of power within the audio
+subsystem at all times. It is independent of other kernel PM and as
+such, can easily co-exist with the other PM systems.
+
+DAPM is also completely transparent to all user space applications as
+all power switching is done within the ASoC core. No code changes or
+recompiling are required for user space applications. DAPM makes power
+switching decisions based upon any audio stream (capture/playback)
+activity and audio mixer settings within the device.
+
+DAPM spans the whole machine. It covers power control within the entire
+audio subsystem, this includes internal codec power blocks and machine
+level power systems.
+
+There are 4 power domains within DAPM
+
+ 1. Codec bias domain - VREF, VMID (core codec and audio power)
+ Usually controlled at codec probe/remove and suspend/resume, although
+ can be set at stream time if power is not needed for sidetone, etc.
+
+ 2. Platform/Machine domain - physically connected inputs and outputs
+ Is platform/machine and user action specific, is configured by the
+ machine driver and responds to asynchronous events e.g when HP
+ are inserted
+
+ 3. Path domain - audio subsystem signal paths
+ Automatically set when mixer and mux settings are changed by the user.
+ e.g. alsamixer, amixer.
+
+ 4. Stream domain - DACs and ADCs.
+ Enabled and disabled when stream playback/capture is started and
+ stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching decisions are made automatically by consulting an audio
+routing map of the whole machine. This map is specific to each machine and
+consists of the interconnections between every audio component (including
+internal codec components). All audio components that effect power are called
+widgets hereafter.
+
+
+2. DAPM Widgets
+===============
+
+Audio DAPM widgets fall into a number of types:-
+
+ o Mixer - Mixes several analog signals into a single analog signal.
+ o Mux - An analog switch that outputs only one of many inputs.
+ o PGA - A programmable gain amplifier or attenuation widget.
+ o ADC - Analog to Digital Converter
+ o DAC - Digital to Analog Converter
+ o Switch - An analog switch
+ o Input - A codec input pin
+ o Output - A codec output pin
+ o Headphone - Headphone (and optional Jack)
+ o Mic - Mic (and optional Jack)
+ o Line - Line Input/Output (and optional Jack)
+ o Speaker - Speaker
+ o Supply - Power or clock supply widget used by other widgets.
+ o Regulator - External regulator that supplies power to audio components.
+ o Clock - External clock that supplies clock to audio components.
+ o AIF IN - Audio Interface Input (with TDM slot mask).
+ o AIF OUT - Audio Interface Output (with TDM slot mask).
+ o Siggen - Signal Generator.
+ o DAI IN - Digital Audio Interface Input.
+ o DAI OUT - Digital Audio Interface Output.
+ o DAI Link - DAI Link between two DAI structures */
+ o Pre - Special PRE widget (exec before all others)
+ o Post - Special POST widget (exec after all others)
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets can be added to the sound card by any of the component driver types.
+There are convenience macros defined in soc-dapm.h that can be used to quickly
+build a list of widgets of the codecs and machines DAPM widgets.
+
+Most widgets have a name, register, shift and invert. Some widgets have extra
+parameters for stream name and kcontrols.
+
+
+2.1 Stream Domain Widgets
+-------------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADCs
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
+
+Stream widgets have the following format:-
+
+SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
+
+NOTE: the stream name must match the corresponding stream name in your codec
+snd_soc_codec_dai.
+
+e.g. stream widgets for HiFi playback and capture
+
+SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
+SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
+
+e.g. stream widgets for AIF
+
+SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+
+2.2 Path Domain Widgets
+-----------------------
+
+Path domain widgets have a ability to control or affect the audio signal or
+audio paths within the audio subsystem. They have the following form:-
+
+SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
+
+Any widget kcontrols can be set using the controls and num_controls members.
+
+e.g. Mixer widget (the kcontrols are declared first)
+
+/* Output Mixer */
+static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
+ ARRAY_SIZE(wm8731_output_mixer_controls)),
+
+If you 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 Machine domain Widgets
+--------------------------
+
+Machine widgets are different from codec widgets in that they don't have a
+codec register bit associated with them. A machine widget is assigned to each
+machine audio component (non codec or DSP) that can be independently
+powered. e.g.
+
+ o Speaker Amp
+ o Microphone Bias
+ o Jack connectors
+
+A machine widget can have an optional call back.
+
+e.g. Jack connector widget for an external Mic that enables Mic Bias
+when the Mic is inserted:-
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
+{
+ gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+
+
+2.4 Codec (BIAS) Domain
+-----------------------
+
+The codec bias power domain has no widgets and is handled by the codecs DAPM
+event handler. This handler is called when the codec powerstate is changed wrt
+to any stream event or by kernel PM events.
+
+
+2.5 Virtual Widgets
+-------------------
+
+Sometimes widgets exist in the codec or machine audio map that don't have any
+corresponding soft power control. In this case it is necessary to create
+a virtual widget - a widget with no control bits e.g.
+
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
+
+This can be used to merge to signal paths together in software.
+
+After all the widgets have been defined, they can then be added to the DAPM
+subsystem individually with a call to snd_soc_dapm_new_control().
+
+
+3. Codec/DSP Widget Interconnections
+====================================
+
+Widgets are connected to each other within the codec, platform and machine by
+audio paths (called interconnections). Each interconnection must be defined in
+order to create a map of all audio paths between widgets.
+
+This is easiest with a diagram of the codec or DSP (and schematic of the machine
+audio system), as it requires joining widgets together via their audio signal
+paths.
+
+e.g., from the WM8731 output mixer (wm8731.c)
+
+The WM8731 output mixer has 3 inputs (sources)
+
+ 1. Line Bypass Input
+ 2. DAC (HiFi playback)
+ 3. Mic Sidetone Input
+
+Each input in this example has a kcontrol associated with it (defined in example
+above) and is connected to the output mixer via its kcontrol name. We can now
+connect the destination widget (wrt audio signal) with its source widgets.
+
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+So we have :-
+
+ Destination Widget <=== Path Name <=== Source Widget
+
+Or:-
+
+ Sink, Path, Source
+
+Or :-
+
+ "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
+
+When there is no path name connecting widgets (e.g. a direct connection) we
+pass NULL for the path name.
+
+Interconnections are created with a call to:-
+
+snd_soc_dapm_connect_input(codec, sink, path, source);
+
+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+interconnections have been registered with the core. This causes the core to
+scan the codec and machine so that the internal DAPM state matches the
+physical state of the machine.
+
+
+3.1 Machine Widget Interconnections
+-----------------------------------
+Machine widget interconnections are created in the same way as codec ones and
+directly connect the codec pins to machine level widgets.
+
+e.g. connects the speaker out codec pins to the internal speaker.
+
+ /* ext speaker connected to codec pins LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+This allows the DAPM to power on and off pins that are connected (and in use)
+and pins that are NC respectively.
+
+
+4 Endpoint Widgets
+===================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+ o Headphone Jack
+ o Internal Speaker
+ o Internal Mic
+ o Mic Jack
+ o Codec Pins
+
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
+
+
+5 DAPM Widget Events
+====================
+
+Some widgets can register their interest with the DAPM core in PM events.
+e.g. A Speaker with an amplifier registers a widget so the amplifier can be
+powered only when the spk is in use.
+
+/* turn speaker amplifier on/off depending on use */
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+ gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
+ SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+
+Please see soc-dapm.h for all other widgets that support events.
+
+
+5.1 Event types
+---------------
+
+The following event types are supported by event widgets.
+
+/* dapm event types */
+#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
+#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
+#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
+#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
+#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
+#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 000000000..fcf82a417
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h. ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+ user visible jack. In embedded systems it is common for multiple
+ to be present on a single jack but handled by separate bits of
+ hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+ automatically based on the detected jack status (eg, turning off the
+ headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone. Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space. The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack. Each snd_soc_jack has zero or more of these
+which are updated automatically. They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins(). The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update. The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function. Other methods are
+also available, for example integrated into CODECs. One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware. The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 000000000..74056dba5
--- /dev/null
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -0,0 +1,93 @@
+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each componnent which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
+
+The machine driver can contain codec and platform specific code. It registers
+the audio subsystem with the kernel as a platform device and is represented by
+the following struct:-
+
+/* SoC machine */
+struct snd_soc_card {
+ char *name;
+
+ ...
+
+ int (*probe)(struct platform_device *pdev);
+ int (*remove)(struct platform_device *pdev);
+
+ /* the pre and post PM functions are used to do any PM work before and
+ * after the codec and DAIs do any PM work. */
+ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
+ int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
+ int (*resume_pre)(struct platform_device *pdev);
+ int (*resume_post)(struct platform_device *pdev);
+
+ ...
+
+ /* CPU <--> Codec DAI links */
+ struct snd_soc_dai_link *dai_link;
+ int num_links;
+
+ ...
+};
+
+probe()/remove()
+----------------
+probe/remove are optional. Do any machine specific probe here.
+
+
+suspend()/resume()
+------------------
+The machine driver has pre and post versions of suspend and resume to take care
+of any machine audio tasks that have to be done before or after the codec, DAIs
+and DMA is suspended and resumed. Optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAIs together. It can
+also be used to set up the DAI system clock and for any machine related DAI
+initialisation e.g. the machine audio map can be connected to the codec audio
+map, unconnected codec pins can be set as such.
+
+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai_name = "pxa-is2-dai",
+ .codec_dai_name = "wm8731-hifi",
+ .platform_name = "pxa-pcm-audio",
+ .codec_name = "wm8713-codec.0-001a",
+ .init = corgi_wm8731_init,
+ .ops = &corgi_ops,
+};
+
+struct snd_soc_card then sets up the machine with its DAIs. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_card snd_soc_corgi = {
+ .name = "Corgi",
+ .dai_link = &corgi_dai,
+ .num_links = 1,
+};
+
+
+Machine Power Map
+-----------------
+
+The machine driver can optionally extend the codec power map and to become an
+audio power map of the audio subsystem. This allows for automatic power up/down
+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
+sockets in the machine init function.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the DAI init function.
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 000000000..ff88f52ee
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,95 @@
+ALSA SoC Layer
+==============
+
+The overall project goal of the ALSA System on Chip (ASoC) layer is to
+provide better ALSA support for embedded system-on-chip processors (e.g.
+pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC
+subsystem there was some support in the kernel for SoC audio, however it
+had some limitations:-
+
+ * Codec drivers were often tightly coupled to the underlying SoC
+ CPU. This is not ideal and leads to code duplication - for example,
+ Linux had different wm8731 drivers for 4 different SoC platforms.
+
+ * There was no standard method to signal user initiated audio events (e.g.
+ Headphone/Mic insertion, Headphone/Mic detection after an insertion
+ event). These are quite common events on portable devices and often require
+ machine specific code to re-route audio, enable amps, etc., after such an
+ event.
+
+ * Drivers tended to power up the entire codec when playing (or
+ recording) audio. This is fine for a PC, but tends to waste a lot of
+ power on portable devices. There was also no support for saving
+ power via changing codec oversampling rates, bias currents, etc.
+
+
+ASoC Design
+===========
+
+The ASoC layer is designed to address these issues and provide the following
+features :-
+
+ * Codec independence. Allows reuse of codec drivers on other platforms
+ and machines.
+
+ * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
+ interface and codec registers its audio interface capabilities with the
+ core and are subsequently matched and configured when the application
+ hardware parameters are known.
+
+ * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+ its minimum power state at all times. This includes powering up/down
+ internal power blocks depending on the internal codec audio routing and any
+ active streams.
+
+ * Pop and click reduction. Pops and clicks can be reduced by powering the
+ codec up/down in the correct sequence (including using digital mute). ASoC
+ signals the codec when to change power states.
+
+ * Machine specific controls: Allow machines to add controls to the sound card
+ (e.g. volume control for speaker amplifier).
+
+To achieve all this, ASoC basically splits an embedded audio system into
+multiple re-usable component drivers :-
+
+ * Codec class drivers: The codec class driver is platform independent and
+ contains audio controls, audio interface capabilities, codec DAPM
+ definition and codec IO functions. This class extends to BT, FM and MODEM
+ ICs if required. Codec class drivers should be generic code that can run
+ on any architecture and machine.
+
+ * Platform class drivers: The platform class driver includes the audio DMA
+ engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
+ and any audio DSP drivers for that platform.
+
+ * Machine class driver: The machine driver class acts as the glue that
+ decribes and binds the other component drivers together to form an ALSA
+ "sound card device". It handles any machine specific controls and
+ machine level audio events (e.g. turning on an amp at start of playback).
+
+
+Documentation
+=============
+
+The documentation is spilt into the following sections:-
+
+overview.txt: This file.
+
+codec.txt: Codec driver internals.
+
+DAI.txt: Description of Digital Audio Interface standards and how to configure
+a DAI within your codec and CPU DAI drivers.
+
+dapm.txt: Dynamic Audio Power Management
+
+platform.txt: Platform audio DMA and DAI.
+
+machine.txt: Machine driver internals.
+
+pop_clicks.txt: How to minimise audio artifacts.
+
+clocking.txt: ASoC clocking for best power performance.
+
+jack.txt: ASoC jack detection.
+
+DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 000000000..3a08a2c91
--- /dev/null
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -0,0 +1,79 @@
+ASoC Platform Driver
+====================
+
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. The platform drivers only target the SoC CPU and must
+have no board specific code.
+
+Audio DMA
+=========
+
+The platform DMA driver optionally supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
+ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+ int (*hw_free)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+ int (*trigger)(struct snd_pcm_substream *, int);
+};
+
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
+
+struct snd_soc_platform_driver {
+ char *name;
+
+ int (*probe)(struct platform_device *pdev);
+ int (*remove)(struct platform_device *pdev);
+ int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+ int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+
+ /* pcm creation and destruction */
+ int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
+ void (*pcm_free)(struct snd_pcm *);
+
+ /*
+ * For platform caused delay reporting.
+ * Optional.
+ */
+ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+ struct snd_soc_dai *);
+
+ /* platform stream ops */
+ struct snd_pcm_ops *pcm_ops;
+};
+
+Please refer to the ALSA driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+An example DMA driver is soc/pxa/pxa2xx-pcm.c
+
+
+SoC DAI Drivers
+===============
+
+Each SoC DAI driver must provide the following features:-
+
+ 1) Digital audio interface (DAI) description
+ 2) Digital audio interface configuration
+ 3) PCM's description
+ 4) SYSCLK configuration
+ 5) Suspend and resume (optional)
+
+Please see codec.txt for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+ 1) DAPM graph
+ 2) Mixer controls
+ 3) DMA IO to/from DSP buffers (if applicable)
+ 4) Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
new file mode 100644
index 000000000..e1e74daa4
--- /dev/null
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is noticeable on PCs when an
+audio module is either loaded or unloaded (at module load time the sound card is
+powered up and causes a popping noise on the speakers).
+
+Pops and clicks can be more frequent on portable systems with DAPM. This is
+because the components within the subsystem are being dynamically powered
+depending on the audio usage and this can subsequently cause a small pop or
+click every time a component power state is changed.
+
+
+Minimising Playback Pops and Clicks
+===================================
+
+Playback pops in portable audio subsystems cannot be completely eliminated
+currently, however future audio codec hardware will have better pop and click
+suppression. Pops can be reduced within playback by powering the audio
+components in a specific order. This order is different for startup and
+shutdown and follows some basic rules:-
+
+ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
+
+ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
+
+This assumes that the codec PCM output path from the DAC is via a mixer and then
+a PGA (programmable gain amplifier) before being output to the speakers.
+
+
+Minimising Capture Pops and Clicks
+==================================
+
+Capture artifacts are somewhat easier to get rid as we can delay activating the
+ADC until all the pops have occurred. This follows similar power rules to
+playback in that components are powered in a sequence depending upon stream
+startup or shutdown.
+
+ Startup Order - Input PGA --> Mixers --> ADC
+
+ Shutdown Order - ADC --> Mixers --> Input PGA
+
+
+Zipper Noise
+============
+An unwanted zipper noise can occur within the audio playback or capture stream
+when a volume control is changed near its maximum gain value. The zipper noise
+is heard when the gain increase or decrease changes the mean audio signal
+amplitude too quickly. It can be minimised by enabling the zero cross setting
+for each volume control. The ZC forces the gain change to occur when the signal
+crosses the zero amplitude line.
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
new file mode 100644
index 000000000..0b191a23f
--- /dev/null
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -0,0 +1,200 @@
+The ALSA API can provide two different system timestamps:
+
+- Trigger_tstamp is the system time snapshot taken when the .trigger
+callback is invoked. This snapshot is taken by the ALSA core in the
+general case, but specific hardware may have synchronization
+capabilities or conversely may only be able to provide a correct
+estimate with a delay. In the latter two cases, the low-level driver
+is responsible for updating the trigger_tstamp at the most appropriate
+and precise moment. Applications should not rely solely on the first
+trigger_tstamp but update their internal calculations if the driver
+provides a refined estimate with a delay.
+
+- tstamp is the current system timestamp updated during the last
+event or application query.
+The difference (tstamp - trigger_tstamp) defines the elapsed time.
+
+The ALSA API provides reports two basic pieces of information, avail
+and delay, which combined with the trigger and current system
+timestamps allow for applications to keep track of the 'fullness' of
+the ring buffer and the amount of queued samples.
+
+The use of these different pointers and time information depends on
+the application needs:
+
+- 'avail' reports how much can be written in the ring buffer
+- 'delay' reports the time it will take to hear a new sample after all
+queued samples have been played out.
+
+When timestamps are enabled, the avail/delay information is reported
+along with a snapshot of system time. Applications can select from
+CLOCK_REALTIME (NTP corrections including going backwards),
+CLOCK_MONOTONIC (NTP corrections but never going backwards),
+CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
+dynamically with sw_params
+
+
+The ALSA API also provide an audio_tstamp which reflects the passage
+of time as measured by different components of audio hardware. In
+ascii-art, this could be represented as follows (for the playback
+case):
+
+
+--------------------------------------------------------------> time
+ ^ ^ ^ ^ ^
+ | | | | |
+ analog link dma app FullBuffer
+ time time time time time
+ | | | | |
+ |< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
+ |<----------------- delay---------------------->| |
+ |<----ring buffer length---->|
+
+The analog time is taken at the last stage of the playback, as close
+as possible to the actual transducer
+
+The link time is taken at the output of the SOC/chipset as the samples
+are pushed on a link. The link time can be directly measured if
+supported in hardware by sample counters or wallclocks (e.g. with
+HDAudio 24MHz or PTP clock for networked solutions) or indirectly
+estimated (e.g. with the frame counter in USB).
+
+The DMA time is measured using counters - typically the least reliable
+of all measurements due to the bursty natured of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query what the hardware supports, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, get an estimate of the timestamp
+accuracy. It can also request the delay-to-analog be included in the
+measurement. Direct access to the link time is very interesting on
+platforms that provide an embedded DSP; measuring directly the link
+time with dedicated hardware, possibly synchronized with system time,
+removes the need to keep track of internal DSP processing times and
+latency.
+
+In case the application requests an audio tstamp that is not supported
+in hardware/low-level driver, the type is overridden as DEFAULT and the
+timestamp will report the DMA time based on the hw_pointer value.
+
+For backwards compatibility with previous implementations that did not
+provide timestamp selection, with a zero-valued COMPAT timestamp type
+the results will default to the HDAudio wall clock for playback
+streams and to the DMA time (hw_ptr) in all other cases.
+
+The audio timestamp accuracy can be returned to user-space, so that
+appropriate decisions are made:
+
+- for dma time (default), the granularity of the transfers can be
+ inferred from the steps between updates and in turn provide
+ information on how much the application pointer can be rewound
+ safely.
+
+- the link time can be used to track long-term drifts between audio
+ and system time using the (tstamp-trigger_tstamp)/audio_tstamp
+ ratio, the precision helps define how much smoothing/low-pass
+ filtering is required. The link time can be either reset on startup
+ or reported as is (the latter being useful to compare progress of
+ different streams - but may require the wallclock to be always
+ running and not wrap-around during idle periods). If supported in
+ hardware, the absolute link time could also be used to define a
+ precise start time (patches WIP)
+
+- including the delay in the audio timestamp may
+ counter-intuitively not increase the precision of timestamps, e.g. if a
+ codec includes variable-latency DSP processing or a chain of
+ hardware components the delay is typically not known with precision.
+
+The accuracy is reported in nanosecond units (using an unsigned 32-bit
+word), which gives a max precision of 4.29s, more than enough for
+audio applications...
+
+Due to the varied nature of timestamping needs, even for a single
+application, the audio_tstamp_config can be changed dynamically. In
+the STATUS ioctl, the parameters are read-only and do not allow for
+any application selection. To work around this limitation without
+impacting legacy applications, a new STATUS_EXT ioctl is introduced
+with read/write parameters. ALSA-lib will be modified to make use of
+STATUS_EXT and effectively deprecate STATUS.
+
+The ALSA API only allows for a single audio timestamp to be reported
+at a time. This is a conscious design decision, reading the audio
+timestamps from hardware registers or from IPC takes time, the more
+timestamps are read the more imprecise the combined measurements
+are. To avoid any interpretation issues, a single (system, audio)
+timestamp is reported. Applications that need different timestamps
+will be required to issue multiple queries and perform an
+interpolation of the results
+
+In some hardware-specific configuration, the system timestamp is
+latched by a low-level audio subsytem, and the information provided
+back to the driver. Due to potential delays in the communication with
+the hardware, there is a risk of misalignment with the avail and delay
+information. To make sure applications are not confused, a
+driver_timestamp field is added in the snd_pcm_status structure; this
+timestamp shows when the information is put together by the driver
+before returning from the STATUS and STATUS_EXT ioctl. in most cases
+this driver_timestamp will be identical to the regular system tstamp.
+
+Examples of typestamping with HDaudio:
+
+1. DMA timestamp, no compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=1
+playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
+playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
+playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
+playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
+playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
+playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
+
+2. DMA timestamp, compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=1 -d
+playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
+playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
+playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
+playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
+playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
+
+3. link timestamp, compensation for DMA+analog delay
+$ ./audio_time -p --ts_type=2 -d
+playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
+playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
+playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
+playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
+playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
+playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
+
+Example 1 shows that the timestamp at the DMA level is close to 1ms
+ahead of the actual playback time (as a side time this sort of
+measurement can help define rewind safeguards). Compensating for the
+DMA-link delay in example 2 helps remove the hardware buffering abut
+the information is still very jittery, with up to one sample of
+error. In example 3 where the timestamps are measured with the link
+wallclock, the timestamps show a monotonic behavior and a lower
+dispersion.
+
+Example 3 and 4 are with USB audio class. Example 3 shows a high
+offset between audio time and system time due to buffering. Example 4
+shows how compensating for the delay exposes a 1ms accuracy (due to
+the use of the frame counter by the driver)
+
+Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
+$ ./audio_time -p -Dhw:1 -t1
+playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
+playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
+playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
+playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
+playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
+playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
+playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
+
+Example 4: DMA timestamp, compensation for delay, delay of ~1ms
+$ ./audio_time -p -Dhw:1 -t1 -d
+playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
+playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
+playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
+playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
+playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
+playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847