Merge remote-tracking branch 'sound/topic/restize-docs' into sound

Bring in the sphinxification of the sound documentation.

14 files changed, 2663 insertions, 0 deletions

diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst
new file mode 100644
index 000000000000..86213234435f
--- /dev/null
+++ b/Documentation/sound/cards/audigy-mixer.rst
@@ -0,0 +1,368 @@
+=============================================
+Sound Blaster Audigy mixer / default DSP code
+=============================================
+
+This is based on sb-live-mixer.rst.
+
+The EMU10K2 chips have a DSP part which can be programmed to support 
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+Digital mixer controls
+======================
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC
+	digital to analog converter
+ADC
+	analog to digital converter
+I2S
+	one-way three wire serial bus for digital sound by Philips Semiconductors
+        (this standard is used for connecting standalone DAC and ADC converters)
+LFE
+	low frequency effects (subwoofer signal)
+AC97
+	a chip containing an analog mixer, DAC and ADC converters
+IEC958
+	S/PDIF
+FX-bus
+	the EMU10K2 chip has an effect bus containing 64 accumulators.
+        Each of the synthesizer voices can feed its output to these accumulators
+        and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+----------------------------------------
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+-------------------------------------------
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+-----------------------------------------
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+--------------------------------------
+This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots 
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+---------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+-----------------------------------
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to 
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+---------------------------------
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+----------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+---------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+---------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+--------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+-----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+-----------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+----------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+-----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+---------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+-------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+----------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+------------------------------------
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+----------------------------------
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+------------------------------------
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+-------------------------------------
+This control is used to attenuate samples for front, surround, center and 
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+-------------------------------------------------
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+PCM stream related controls
+===========================
+
+name='EMU10K1 PCM Volume',index 0-31
+------------------------------------
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+* 0 - mono, default 0xffff (no attenuation)
+* 1 - left, default 0xffff (no attenuation)
+* 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+------------------------------------------
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+* 0 -  mono, A destination (FX-bus 0-63), default 0
+* 1 -  mono, B destination (FX-bus 0-63), default 1
+* 2 -  mono, C destination (FX-bus 0-63), default 2
+* 3 -  mono, D destination (FX-bus 0-63), default 3
+* 4 -  mono, E destination (FX-bus 0-63), default 0
+* 5 -  mono, F destination (FX-bus 0-63), default 0
+* 6 -  mono, G destination (FX-bus 0-63), default 0
+* 7 -  mono, H destination (FX-bus 0-63), default 0
+* 8 -  left, A destination (FX-bus 0-63), default 0
+* 9 -  left, B destination (FX-bus 0-63), default 1
+* 10 -  left, C destination (FX-bus 0-63), default 2
+* 11 -  left, D destination (FX-bus 0-63), default 3
+* 12 -  left, E destination (FX-bus 0-63), default 0
+* 13 -  left, F destination (FX-bus 0-63), default 0
+* 14 -  left, G destination (FX-bus 0-63), default 0
+* 15 -  left, H destination (FX-bus 0-63), default 0
+* 16 - right, A destination (FX-bus 0-63), default 0
+* 17 - right, B destination (FX-bus 0-63), default 1
+* 18 - right, C destination (FX-bus 0-63), default 2
+* 19 - right, D destination (FX-bus 0-63), default 3
+* 20 - right, E destination (FX-bus 0-63), default 0
+* 21 - right, F destination (FX-bus 0-63), default 0
+* 22 - right, G destination (FX-bus 0-63), default 0
+* 23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+name='EMU10K1 PCM Send Volume',index 0-31
+-----------------------------------------
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+*  0 -  mono, A destination attn, default 255 (no attenuation)
+*  1 -  mono, B destination attn, default 255 (no attenuation)
+*  2 -  mono, C destination attn, default 0 (mute)
+*  3 -  mono, D destination attn, default 0 (mute)
+*  4 -  mono, E destination attn, default 0 (mute)
+*  5 -  mono, F destination attn, default 0 (mute)
+*  6 -  mono, G destination attn, default 0 (mute)
+*  7 -  mono, H destination attn, default 0 (mute)
+*  8 -  left, A destination attn, default 255 (no attenuation)
+*  9 -  left, B destination attn, default 0 (mute)
+* 10 -  left, C destination attn, default 0 (mute)
+* 11 -  left, D destination attn, default 0 (mute)
+* 12 -  left, E destination attn, default 0 (mute)
+* 13 -  left, F destination attn, default 0 (mute)
+* 14 -  left, G destination attn, default 0 (mute)
+* 15 -  left, H destination attn, default 0 (mute)
+* 16 - right, A destination attn, default 0 (mute)
+* 17 - right, B destination attn, default 255 (no attenuation)
+* 18 - right, C destination attn, default 0 (mute)
+* 19 - right, D destination attn, default 0 (mute)
+* 20 - right, E destination attn, default 0 (mute)
+* 21 - right, F destination attn, default 0 (mute)
+* 22 - right, G destination attn, default 0 (mute)
+* 23 - right, H destination attn, default 0 (mute)
+
+
+
+MANUALS/PATENTS
+===============
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+LM4545.pdf
+	AC97 Codec
+
+m2049.pdf
+	The EMU10K1 Digital Audio Processor
+
+hog63.ps
+	FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+
+WO 9901813 (A1)
+	Audio Effects Processor with multiple asynchronous streams
+	(Jan. 14, 1999)
+
+WO 9901814 (A1)
+	Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+WO 9901953 (A1)
+	Audio Effects Processor having Decoupled Instruction
+        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (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
+        with a multiport memory onto which multiple asynchronous
+        digital sound samples can be concurrently loaded
+	(Jul. 27, 1999)
+
+US 5930158
+	Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+US 6032235
+	Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+US 6138207
+	Interpolation looping of audio samples in cache connected to
+        system bus with prioritization and modification of bus transfers
+        in accordance with loop ends and minimum block sizes
+	(Oct. 24, 2000)
+
+US 6151670
+	Method for conserving memory storage using a
+        pool of  short term memory registers
+	(Nov. 21, 2000)
+
+US 6195715
+	Interrupt control for multiple programs communicating with
+        a common interrupt by associating programs to GP registers,
+        defining interrupt register, polling GP registers, and invoking
+        callback routine associated with defined interrupt register
+	(Feb. 27, 2001)
diff --git a/Documentation/sound/cards/audiophile-usb.rst b/Documentation/sound/cards/audiophile-usb.rst
new file mode 100644
index 000000000000..a7bb5648331f
--- /dev/null
+++ b/Documentation/sound/cards/audiophile-usb.rst
@@ -0,0 +1,550 @@
+========================================================
+Guide to using M-Audio Audiophile USB with ALSA and Jack
+========================================================
+
+v1.5
+
+Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with 
+ALSA and JACK.
+
+History
+=======
+
+* v1.4 - Thibault Le Meur (2007-07-11)
+
+  - Added Low Endianness nature of 16bits-modes
+    found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+  - Modifying document structure
+
+* v1.5 - Thibault Le Meur (2007-07-12)
+  - Added AC3/DTS passthru info
+
+
+Audiophile USB Specs and correct usage
+======================================
+
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+
+ * Analog Stereo Input (Ai)
+
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
+
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following characteristics:
+
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two interfaces can't use different sample depths at the same time.
+
+Moreover, the Audiophile USB documentation gives the following Warning:
+  Please exit any audio application running before switching between bit depths
+
+Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
+activated at the same time depending on the audio mode selected:
+
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
+
+   - Ai+Ao+Di+Do
+
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out, 
+   or 2 channels in + 4 channels out
+
+   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
+
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
+
+   - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+
+ * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, 
+   though I haven't tested it under Linux
+
+   - Note that in this setup only the Do interface can be enabled
+
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+   to synchronize the device to an external sample clock
+
+   - As a consequence, the Di port must be enable only if an active Digital 
+     source is connected
+   - Enabling Di when no digital source is connected can result in a 
+     synchronization error (for instance sound played at an odd sample rate)
+
+
+Audiophile USB MIDI support in ALSA
+===================================
+
+The Audiophile USB MIDI ports will be automatically supported once the
+following modules have been loaded:
+
+ * snd-usb-audio
+ * snd-seq-midi
+
+No additional setting is required.
+
+
+Audiophile USB Audio support in ALSA
+====================================
+
+Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
+module. This module can work in a default mode (without any device-specific 
+parameter), or in an "advanced" mode with the device-specific parameter called 
+``device_setup``.
+
+Default Alsa driver mode
+------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device 
+capabilities at startup and activate the required mode when required 
+by the applications: for instance if the user is recording in a 
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample 
+rates/depths automatically according to the user's needs. However, those who 
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this 
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now.  In this case the 
+Audiophile interfaces are mapped to alsa pcm devices in the following 
+way (I suppose the device's index is 1):
+
+ * hw:1,0 is Ao in playback and Di in capture
+ * hw:1,1 is Do in playback and Ai in capture
+ * hw:1,2 is Do in AC3/DTS passthrough mode
+
+In this mode, the device uses Big Endian byte-encoding so that 
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for 
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian 
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
+is reported to be big endian in this default driver mode.
+
+Examples:
+
+ * playing a S24_3BE encoded raw file to the Ao port::
+
+   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+
+ * recording a  S24_3BE encoded raw file from the Ai port::
+
+   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
+
+ * playing a S16_BE encoded raw file to the Do port::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+
+ * playing an ac3 sample file to the Do port::
+
+   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
+
+If you're happy with the default Alsa driver mode and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+Advanced module setup
+---------------------
+
+Due to the hardware constraints described above, the device initialization made 
+by the Alsa driver in default mode may result in a corrupted state of the 
+device. For instance, a particularly annoying issue is that the sound captured 
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called ``device_setup`` (this parameter was introduced in kernel
+release 2.6.17)
+
+Initializing the working mode of the Audiophile USB
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+As far as the Audiophile USB device is concerned, this value let the user 
+specify:
+
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not 
+
+When initialized with ``device_setup=0x00``, the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default 
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+16-bit modes
+~~~~~~~~~~~~
+
+The two supported modes are:
+
+ * ``device_setup=0x01``
+
+   - 16bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * ``device_setup=0x11``
+
+   - 16bits 48kHz mode with Di enabled
+   - Ai,Ao,Di,Do can be used at the same time
+   - hw:1,0 is available in capture mode
+   - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+
+24-bit modes
+~~~~~~~~~~~~
+
+The three supported modes are:
+
+ * ``device_setup=0x09``
+
+   - 24bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * ``device_setup=0x19``
+
+   - 24bits 48kHz mode with Di enabled
+   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in capture mode and an active digital source must be 
+     connected to Di
+   - hw:1,2 is not available
+
+ * ``device_setup=0x0D`` or ``0x10``
+
+   - 24bits 96kHz mode
+   - Di is enabled by default for this mode but does not need to be connected 
+     to an active source
+   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in captured mode
+   - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
+mode" above for an aplay command example)
+
+AC3 w/ DTS passthru mode
+~~~~~~~~~~~~~~~~~~~~~~~~
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
+ * ``device_setup=0x03``
+
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru
+   - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+::
+
+   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+How to use the ``device_setup`` parameter
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+The parameter can be given:
+
+ * By manually probing the device (as root):::
+
+   # modprobe -r snd-usb-audio
+   # modprobe snd-usb-audio index=1 device_setup=0x09
+
+ * Or while configuring the modules options in your modules configuration file
+   (typically a .conf file in /etc/modprobe.d/ directory:::
+
+       alias snd-card-1 snd-usb-audio
+       options snd-usb-audio index=1 device_setup=0x09
+
+CAUTION when initializing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+   the module BEFORE the device is turned on. So, if you use the "manual probing"
+   method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead to a misconfiguration of the device. In this case
+   turn off the device, unprobe the snd-usb-audio module, then probe it again with
+   correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+   to  another mode (possibly with another sample-depth), please use also the following 
+   procedure:
+
+   - first turn off the device
+   - de-register the snd-usb-audio module (modprobe -r)
+   - change the device_setup parameter by changing the device_setup
+     option in ``/etc/modprobe.d/*.conf``
+   - turn on the device
+
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+   be enough to ensure the 'stability' of the device initialization.
+
+Technical details for hackers
+-----------------------------
+
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+Audiophile USB's ``device_setup`` structure
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+If you want to understand the device_setup magic numbers for the Audiophile 
+USB, you need some very basic understanding of binary computation. However, 
+this is not required to use the parameter and you may skip this section.
+
+The device_setup is one byte long and its structure is the following:
+::
+
+       +---+---+---+---+---+---+---+---+
+       | b7| b6| b5| b4| b3| b2| b1| b0|
+       +---+---+---+---+---+---+---+---+
+       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+       +---+---+---+---+---+---+---+---+
+
+Where:
+
+ * b0 is the ``SET`` bit
+
+   - it MUST be set if device_setup is initialized 
+
+ * b1 is the ``DTS`` bit
+
+   - it is set only for Digital output with DTS/AC3
+   - this setup is not tested
+
+ * b2 is the Rate selection flag
+
+   - When set to ``1`` the rate range is 48.1-96kHz
+   - Otherwise the sample rate range is 8-48kHz
+
+ * b3 is the bit depth selection flag
+
+   - When set to ``1`` samples are 24bits long
+   - Otherwise they are 16bits long
+   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
+     samples
+
+ * b4 is the Digital input flag
+
+   - When set to ``1`` the device assumes that an active digital source is 
+     connected 
+   - You shouldn't enable Di if no source is seen on the port (this leads to 
+     synchronization issues)
+   - b4 is implied by b2 (since only one port is enabled at a time no synch 
+     error can occur) 
+
+ * b5 to b7 are reserved for future uses, and must be set to ``0``
+
+   - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+
+ * there is no check on the value you will give to device_setup
+
+   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
+     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+
+ * Hardware constraints due to the USB bus limitation aren't checked
+
+   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
+     only be able to use one at the same time
+
+USB implementation details for this device
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+You may safely skip this section if you're not interested in driver 
+hacking.
+
+This section describes some internal aspects of the device and summarizes the 
+data I got by usb-snooping the windows and Linux drivers.
+
+The M-Audio Audiophile USB has 7 USB Interfaces:
+a "USB interface":
+
+ * USB Interface nb.0
+ * USB Interface nb.1
+
+   - Audio Control function
+
+ * USB Interface nb.2
+
+   - Analog Output
+
+ * USB Interface nb.3
+
+   - Digital Output
+
+ * USB Interface nb.4
+
+   - Analog Input
+
+ * USB Interface nb.5
+
+   - Digital Input
+
+ * USB Interface nb.6
+
+   - MIDI interface compliant with the MIDIMAN quirk 
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+
+ * Interface 3 (Digital Out) has an extra Alset nb.6 
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5 
+
+Here is a short description of the AltSettings capabilities:
+
+* AltSettings 1 corresponds to
+
+  - 24-bit depth, 48.1-96kHz sample mode
+  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+
+* AltSettings 2 corresponds to
+
+  - 24-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+
+* AltSettings 3 corresponds to
+
+  - 24-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+
+* AltSettings 4 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+
+* AltSettings 5 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+
+* AltSettings 6 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct initialization of the device, the driver 
+*must* *know* how the device will be used:
+
+ * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
+   registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+   Digital input is connected, and only AltSet nb.3 if Digital input
+   is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+   Digital input is connected, and only AltSet nb.5 if Digital input
+   is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the 
+parse_audio_endpoints function uses a quirk called 
+``audiophile_skip_setting_quirk`` in order to prevent AltSettings not 
+corresponding to device_setup from being registered in the driver.
+
+Audiophile USB and Jack support
+===============================
+
+This section deals with support of the Audiophile USB device in Jack.
+
+There are 2 main potential issues when using Jackd with the device:
+
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+Direct support in Jackd
+-----------------------
+
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember 
+exactly when this support was released into jackd, let's just say that
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
+16bits Big-Endian devices, but since you've read carefully the above
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+::
+
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+Using Alsa plughw
+-----------------
+
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa ``plug`` converter.
+
+For instance here is one way to run Jack with 2 playback channels on Ao and 2 
+capture channels from Ai:
+::
+
+  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
+
+However you may see the following warning message:
+  You appear to be using the ALSA software "plug" layer, probably a result of 
+  using the "default" ALSA device. This is less efficient than it could be. 
+  Consider using a hardware device instead rather than using the plug layer.
+
+Getting 2 input and/or output interfaces in Jack
+------------------------------------------------
+
+As you can see, starting the Jack server this way will only enable 1 stereo
+input (Di or Ai) and 1 stereo output (Ao or Do).
+
+This is due to the following restrictions:
+
+* Jack can only open one capture device and one playback device at a time
+* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
+  (and optionally hw:1,2)
+
+If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
+combine the Alsa devices into one logical "complex" device.
+
+If you want to give it a try, I recommend reading the information from
+this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
+It is related to another device (ice1712) but can be adapted to suit
+the Audiophile USB.
+
+Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
+
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
+  file 
+* start jackd with this device
+
+I had no success in testing this for now, if you have any success with this kind 
+of setup, please drop me an email.
diff --git a/Documentation/sound/cards/bt87x.rst b/Documentation/sound/cards/bt87x.rst
new file mode 100644
index 000000000000..912732d3ef9e
--- /dev/null
+++ b/Documentation/sound/cards/bt87x.rst
@@ -0,0 +1,83 @@
+=================
+ALSA BT87x Driver
+=================
+
+Intro
+=====
+
+You might have noticed that the bt878 grabber cards have actually
+*two* PCI functions:
+::
+
+  $ lspci
+  [ ... ]
+  00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+  00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+  [ ... ]
+
+The first does video, it is backward compatible to the bt848.  The second
+does audio.  snd-bt87x is a driver for the second function.  It's a sound
+driver which can be used for recording sound (and *only* recording, no
+playback).  As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+Some cards do not bother to connect anything to the audio input pins of
+the chip, and some other cards use the audio function to transport MPEG
+video data, so it's quite possible that audio recording may not work
+with your card.
+
+
+Driver Status
+=============
+
+The driver is now stable.  However, it doesn't know about many TV cards,
+and it refuses to load for cards it doesn't know.
+
+If the driver complains ("Unknown TV card found, the audio driver will
+not load"), you can specify the ``load_all=1`` option to force the driver to
+try to use the audio capture function of your card.  If the frequency of
+recorded data is not right, try to specify the ``digital_rate`` option with
+other values than the default 32000 (often it's 44100 or 64000).
+
+If you have an unknown card, please mail the ID and board name to
+<alsa-devel@alsa-project.org>, regardless of whether audio capture works
+or not, so that future versions of this driver know about your card.
+
+
+Audio modes
+===========
+
+The chip knows two different modes (digital/analog).  snd-bt87x
+registers two PCM devices, one for each mode.  They cannot be used at
+the same time.
+
+
+Digital audio mode
+==================
+
+The first device (hw:X,0) gives you 16 bit stereo sound.  The sample
+rate depends on the external source which feeds the Bt87x with digital
+sound via I2S interface.
+
+
+Analog audio mode (A/D)
+=======================
+
+The second device (hw:X,1) gives you 8 or 16 bit mono sound.  Supported
+sample rates are between 119466 and 448000 Hz (yes, these numbers are
+that high).  If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the
+maximum sample rate is 1792000 Hz, but audio data becomes unusable
+beyond 896000 Hz on my card.
+
+The chip has three analog inputs.  Consequently you'll get a mixer
+device to control these.
+
+
+Have fun,
+
+  Clemens
+
+
+Written by Clemens Ladisch <clemens@ladisch.de>
+big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org>
diff --git a/Documentation/sound/cards/cmipci.rst b/Documentation/sound/cards/cmipci.rst
new file mode 100644
index 000000000000..9ea1de6ec4ce
--- /dev/null
+++ b/Documentation/sound/cards/cmipci.rst
@@ -0,0 +1,272 @@
+=================================================
+Brief Notes on C-Media 8338/8738/8768/8770 Driver
+=================================================
+
+Takashi Iwai <tiwai@suse.de>
+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks.  Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight differences between the two DACs:
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+  supports only S16LE.
+- The second DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled.  Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+.. WARNING::
+  When "Four Channel Mode" switch is off, the output from rear speakers
+  will be FULL VOLUME regardless of Master and PCM volumes [#]_.
+  This might damage your audio equipment.  Please disconnect speakers
+  before your turn off this switch.
+
+
+.. [#]
+  Well.. I once got the output with correct volume (i.e. same with the
+  front one) and was so excited.  It was even with "Four Channel" bit
+  on and "double DAC" mode.  Actually I could hear separate 4 channels
+  from front and rear speakers!  But.. after reboot, all was gone.
+  It's a very pity that I didn't save the register dump at that
+  time..  Maybe there is an unknown register to achieve this...
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default.  If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else.  When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function.  This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6".  You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
+4) channels.  While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels.  Since the multi-channel playback mode uses both DACs, you
+cannot operate with full-duplex.
+
+The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
+in alsa-lib.  For example, you can play a WAV file with 6 channels like
+::
+
+	% aplay -Dsurround51 sixchannels.wav
+
+For programming the 4/6 channel playback, you need to specify the PCM
+channels as you like and set the format S16LE.  For example, for playback
+with 4 channels,
+::
+
+	snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+	    // or mmap if you like
+	snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+	snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+and use the interleaved 4 channel data.
+
+There are some control switches affecting to the speaker connections:
+
+Line-In Mode
+	an enum control to change the behavior of line-in
+	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
+	be selected.  The last item is available only with model 039
+	or newer. 
+	When "Rear Output" is chosen, the surround channels 3 and 4
+	are output to line-in jack.
+Mic-In Mode
+	an enum control to change the behavior of mic-in
+	jack.  Either "Mic-In" or "Center/LFE Output" can be
+	selected. 
+	When "Center/LFE Output" is chosen, the center and bass
+	channels (channels 5 and 6) are output to mic-in jack. 
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very cheap
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2).  Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+::
+
+	% aplay -Dhw:0,2 foo.wav
+
+or
+
+::
+
+	% aplay -Dspdif foo.wav
+
+24bit format is also supported experimentally.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+respectively, so you cannot playback both analog and digital streams
+simultaneously.
+
+To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+control via mixer or alsactl ("IEC958" is the official name of
+so-called S/PDIF).  Then you'll see the red light on from the card so
+you know that's working obviously :)
+The SPDIF input is always enabled, so you can hear SPDIF input data
+from line-out with "IEC958 In Monitor" switch at any time (see
+below).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used.  Otherwise it's turned
+off.  (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+IEC958 Mix Analog
+	Mix analog PCM playback and FM-OPL/3 streams and
+	output through SPDIF.  This switch appears only on old chip
+	models (CM8738 033 and 037).
+
+	Note: without this control you can output PCM to SPDIF.
+	This is "mixing" of streams, so e.g. it's not for AC3 output
+	(see the next section).
+
+IEC958 In Select
+	Select SPDIF input, the internal CD-in (false)
+	and the external input (true).
+
+IEC958 Loop
+	SPDIF input data is loop back into SPDIF
+	output (aka bypass)
+
+IEC958 Copyright
+	Set the copyright bit.
+
+IEC958 5V
+	Select 0.5V (coax) or 5V (optical) interface.
+	On some cards this doesn't work and you need to change the
+	configuration with hardware dip-switch.
+
+IEC958 In Monitor
+	SPDIF input is routed to DAC.
+
+IEC958 In Phase Inverse
+	Set SPDIF input format as inverse.
+	[FIXME: this doesn't work on all chips..]
+
+IEC958 In Valid
+	Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF.  This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib.  See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop".  The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+On the model 033, AC3 is implemented by the software conversion in
+the alsa-lib.  If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option.  This doesn't matter
+on the newer models.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes.  Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+MIDI CONTROLLER
+---------------
+
+With CMI8338 chips, the MPU401-UART interface is disabled as default.
+You need to set the module option "mpu_port" to a valid I/O port address
+to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
+0x330.  Choose a value that doesn't conflict with other cards.
+
+With CMI8738 and newer chips, the MIDI interface is enabled by default
+and the driver automatically chooses a port address.
+
+There is *no* hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation.  On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+I don't know why..
+
+CMI8768 and newer chips do not have the FM synth.
+
+
+Joystick and Modem
+------------------
+
+The legacy joystick is supported.  To enable the joystick support, pass
+joystick_port=1 module option.  The value 1 means the auto-detection.
+If the auto-detection fails, try to pass the exact I/O address.
+
+The modem is enabled dynamically via a card control switch "Modem".
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci.  If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff --git a/Documentation/sound/cards/emu10k1-jack.rst b/Documentation/sound/cards/emu10k1-jack.rst
new file mode 100644
index 000000000000..6597f1ea83f0
--- /dev/null
+++ b/Documentation/sound/cards/emu10k1-jack.rst
@@ -0,0 +1,78 @@
+=================================================================
+Low latency, multichannel audio with JACK and the emu10k1/emu10k2
+=================================================================
+
+This document is a guide to using the emu10k1 based devices with JACK for low
+latency, multichannel recording functionality.  All of my recent work to allow
+Linux users to use the full capabilities of their hardware has been inspired 
+by the kX Project.  Without their work I never would have discovered the true
+power of this hardware.
+
+	http://www.kxproject.com
+						- Lee Revell, 2005.03.30
+
+
+Until recently, emu10k1 users on Linux did not have access to the same low
+latency, multichannel features offered by the "kX ASIO" feature of their
+Windows driver.  As of ALSA 1.0.9 this is no more!
+
+For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
+channels.  With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
+even 32 (0.66ms) frames should work well.
+
+The configuration is slightly more involved than on Windows, as you have to
+select the correct device for JACK to use.  Actually, for qjackctl users it's
+fairly self explanatory - select Duplex, then for capture and playback select
+the multichannel devices, set the in and out channels to 16, and the sample
+rate to 48000Hz.  The command line looks like this:
+::
+
+  /usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
+
+This will give you 16 input ports and 16 output ports.
+
+The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
+Audigy).  The mapping from FX bus to physical output is described in
+sb-live-mixer.rst (or audigy-mixer.rst).
+
+The 16 input ports are connected to the 16 physical inputs.  Contrary to
+popular belief, all emu10k1 cards are multichannel cards.  Which of these
+input channels have physical inputs connected to them depends on the card
+model.  Trial and error is highly recommended; the pinout diagrams
+for the card have been reverse engineered by some enterprising kX users and are 
+available on the internet.  Meterbridge is helpful here, and the kX forums are
+packed with useful information.
+
+Each input port will either correspond to a digital (SPDIF) input, an analog
+input, or nothing.  The one exception is the SBLive! 5.1.  On these devices,
+the second and third input ports are wired to the center/LFE output.  You will
+still see 16 capture channels, but only 14 are available for recording inputs.
+
+This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
+ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
+channels.
+
+JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
+
+==============  ========        ============
+JACK		Epilog		FXBUS2(nr)
+==============  ========        ============
+capture_1	asio14		FXBUS2(0xe)
+capture_2	asio15		FXBUS2(0xf)
+capture_3	asio0		FXBUS2(0x0)	
+~capture_4	Center		EXTOUT(0x11)	// mapped to by Center
+~capture_5	LFE		EXTOUT(0x12)	// mapped to by LFE
+capture_6	asio3		FXBUS2(0x3)
+capture_7	asio4		FXBUS2(0x4)
+capture_8	asio5		FXBUS2(0x5)
+capture_9	asio6		FXBUS2(0x6)
+capture_10	asio7		FXBUS2(0x7)
+capture_11	asio8		FXBUS2(0x8)
+capture_12	asio9		FXBUS2(0x9)
+capture_13	asio10		FXBUS2(0xa)
+capture_14	asio11		FXBUS2(0xb)
+capture_15	asio12		FXBUS2(0xc)
+capture_16	asio13		FXBUS2(0xd)
+==============  ========        ============
+
+TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/cards/hdspm.rst b/Documentation/sound/cards/hdspm.rst
new file mode 100644
index 000000000000..5373e51ed076
--- /dev/null
+++ b/Documentation/sound/cards/hdspm.rst
@@ -0,0 +1,379 @@
+=======================================
+Software Interface ALSA-DSP MADI Driver 
+=======================================
+
+(translated from German, so no good English ;-), 
+
+2004 - winfried ritsch
+
+
+Full functionality has been added to the driver. Since some of
+the Controls and startup-options  are ALSA-Standard and only the
+special Controls are described and discussed below.
+
+
+Hardware functionality
+======================
+   
+Audio transmission
+------------------
+
+* number of channels --  depends on transmission mode
+
+		The number of channels chosen is from 1..Nmax. The reason to
+		use for a lower number of channels is only resource allocation,
+		since unused DMA channels are disabled and less memory is
+		allocated. So also the throughput of the PCI system can be
+		scaled. (Only important for low performance boards).
+
+* Single Speed -- 1..64 channels 
+
+.. note::
+		 (Note: Choosing the 56channel mode for transmission or as
+		 receiver, only 56 are transmitted/received over the MADI, but
+		 all 64 channels are available for the mixer, so channel count
+		 for the driver)
+
+* Double Speed -- 1..32 channels
+
+.. note::
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 28 are transmitted/received
+		 over the MADI, but all 32 channels are available for the mixer,
+		 so channel count for the driver
+
+
+* Quad Speed -- 1..16 channels 
+
+.. note::
+		 Choosing the 56-channel mode for
+		 transmission/receive-mode , only 14 are transmitted/received
+		 over the MADI, but all 16 channels are available for the mixer,
+		 so channel count for the driver
+
+* Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+* Sample Rates --
+
+       Single Speed -- 32000, 44100, 48000
+
+       Double Speed -- 64000, 88200, 96000 (untested)
+
+       Quad Speed -- 128000, 176400, 192000 (untested)
+
+* access-mode -- MMAP (memory mapped), Not interleaved (PCM_NON-INTERLEAVED)
+
+* buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+* fragments -- 2
+
+* Hardware-pointer -- 2 Modi
+
+
+		 The Card supports the readout of the actual Buffer-pointer,
+		 where DMA reads/writes. Since of the bulk mode of PCI it is only
+		 64 Byte accurate. SO it is not really usable for the
+		 ALSA-mid-level functions (here the buffer-ID gives a better
+		 result), but if MMAP is used by the application. Therefore it
+		 can be configured at load-time with the parameter
+		 precise-pointer.
+
+
+.. hint::
+		 (Hint: Experimenting I found that the pointer is maximum 64 to
+		 large never to small. So if you subtract 64 you always have a
+		 safe pointer for writing, which is used on this mode inside
+		 ALSA. In theory now you can get now a latency as low as 16
+		 Samples, which is a quarter of the interrupt possibilities.)
+
+   * Precise Pointer -- off
+					interrupt used for pointer-calculation
+				
+   * Precise Pointer -- on
+					hardware pointer used.
+
+Controller
+----------
+
+Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+use the standard mixer-controls, since this would break most of
+(especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+provided by a 2-dimensional controller using the
+hwdep-interface. 
+
+Also all 128+256 Peak and RMS-Meter can be accessed via the
+hwdep-interface. Since it could be a performance problem always
+copying and converting Peak and RMS-Levels even if you just need
+one, I decided to export the hardware structure, so that of
+needed some driver-guru can implement a memory-mapping of mixer
+or peak-meters over ioctl, or also to do only copying and no
+conversion. A test-application shows the usage of the controller.
+
+* Latency Controls --- not implemented !!!
+
+.. note::
+	   Note: Within the windows-driver the latency is accessible of a
+	   control-panel, but buffer-sizes are controlled with ALSA from
+	   hwparams-calls and should not be changed in run-state, I did not
+	   implement it here.
+
+
+* System Clock -- suspended !!!!
+
+  * Name -- "System Clock Mode"
+
+  * Access -- Read Write
+    
+  * Values -- "Master" "Slave"
+
+.. note::
+		  !!!! This is a hardware-function but is in conflict with the
+		  Clock-source controller, which is a kind of ALSA-standard. I
+		  makes sense to set the card to a special mode (master at some
+		  frequency or slave), since even not using an Audio-application
+		  a studio should have working synchronisations setup. So use
+		  Clock-source-controller instead !!!!
+
+* Clock Source  
+
+  * Name -- "Sample Clock Source"
+
+  * Access -- Read Write
+
+  * Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+    "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+    "Internal 96.0 kHz"
+
+		 Choose between Master at a specific Frequency and so also the
+		 Speed-mode or Slave (Autosync). Also see  "Preferred Sync Ref"
+
+.. warning::
+       !!!! This is no pure hardware function but was implemented by
+       ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+* Preferred Sync Ref
+
+  * Name -- "Preferred Sync Reference"
+
+  * Access -- Read Write
+
+  * Values -- "Word" "MADI"
+
+
+		 Within the Auto-sync-Mode the preferred Sync Source can be
+		 chosen. If it is not available another is used if possible.
+
+.. note::
+		 Note: Since MADI has a much higher bit-rate than word-clock, the
+		 card should synchronise better in MADI Mode. But since the
+		 RME-PLL is very good, there are almost no problems with
+		 word-clock too. I never found a difference.
+
+
+* TX 64 channel
+
+  * Name -- "TX 64 channels mode"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+		 Using 64-channel-modus (1) or 56-channel-modus for
+		 MADI-transmission (0).
+
+
+.. note::
+		 Note: This control is for output only. Input-mode is detected
+		 automatically from hardware sending MADI.
+
+
+* Clear TMS
+
+  * Name -- "Clear Track Marker"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+
+		 Don't use to lower 5 Audio-bits on AES as additional Bits.
+        
+
+* Safe Mode oder Auto Input
+
+  * Name -- "Safe Mode"
+
+  * Access -- Read Write
+
+  * Values -- 0 1 (default on)
+
+		 If on (1), then if either the optical or coaxial connection
+		 has a failure, there is a takeover to the working one, with no
+		 sample failure. Its only useful if you use the second as a
+		 backup connection.
+
+* Input
+
+  * Name -- "Input Select"
+
+  * Access -- Read Write
+
+  * Values -- optical coaxial
+
+
+		 Choosing the Input, optical or coaxial. If Safe-mode is active,
+		 this is the preferred Input.
+
+Mixer
+-----
+
+* Mixer
+
+  * Name -- "Mixer"
+
+  * Access -- Read Write
+
+  * Values - <channel-number 0-127> <Value 0-65535>
+
+
+		 Here as a first value the channel-index is taken to get/set the
+		 corresponding mixer channel, where 0-63 are the input to output
+		 fader and 64-127 the playback to outputs fader. Value 0
+		 is channel muted 0 and 32768 an amplification of  1.
+
+* Chn 1-64
+
+       fast mixer for the ALSA-mixer utils. The diagonal of the
+       mixer-matrix is implemented from playback to output.
+       
+
+* Line Out
+
+  * Name  -- "Line Out"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+		 Switching on and off the analog out, which has nothing to do
+		 with mixing or routing. the analog outs reflects channel 63,64.
+
+
+Information (only read access)
+------------------------------
+ 
+* Sample Rate
+
+  * Name -- "System Sample Rate"
+
+  * Access -- Read-only
+
+		 getting the sample rate.
+
+
+* External Rate measured
+
+  * Name -- "External Rate"
+
+  * Access -- Read only
+
+
+		 Should be "Autosync Rate", but Name used is
+		 ALSA-Scheme. External Sample frequency liked used on Autosync is
+		 reported.
+
+
+* MADI Sync Status
+
+  * Name -- "MADI Sync Lock Status"
+
+  * Access -- Read
+
+  * Values -- 0,1,2
+
+       MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+* Word Clock Sync Status
+
+  * Name -- "Word Clock Lock Status"
+
+  * Access -- Read
+
+  * Values -- 0,1,2
+
+       Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+* AutoSync
+
+  * Name -- "AutoSync Reference"
+
+  * Access -- Read
+
+  * Values -- "WordClock", "MADI", "None"
+
+		 Sync-Reference is either "WordClock", "MADI" or none.
+
+* RX 64ch --- noch nicht implementiert
+
+       MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+* AB_inp   --- not tested 
+
+		 Used input for Auto-Input.
+
+
+* actual Buffer Position --- not implemented
+
+	   !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter
+=================
+
+* index int array (min = 1, max = 8) 
+
+     Index value for RME HDSPM interface. card-index within ALSA
+
+     note: ALSA-standard
+
+* id string array (min = 1, max = 8) 
+
+     ID string for RME HDSPM interface.
+
+     note: ALSA-standard
+
+* enable int array (min = 1, max = 8)
+
+     Enable/disable specific HDSPM sound-cards.
+
+     note: ALSA-standard
+
+* precise_ptr int array (min = 1, max = 8)
+
+     Enable precise pointer, or disable.
+
+.. note::
+     note: Use only when the application supports this (which is a special case).
+
+* line_outs_monitor int array (min = 1, max = 8)
+
+     Send playback streams to analog outs by default.
+
+.. note::
+	  note: each playback channel is mixed to the same numbered output
+	  channel (routed). This is against the ALSA-convention, where all
+	  channels have to be muted on after loading the driver, but was
+	  used before on other cards, so i historically use it again)
+
+
+
+* enable_monitor int array (min = 1, max = 8)
+
+     Enable Analog Out on Channel 63/64 by default.
+
+.. note ::
+      note: here the analog output is enabled (but not routed).
diff --git a/Documentation/sound/cards/img-spdif-in.rst b/Documentation/sound/cards/img-spdif-in.rst
new file mode 100644
index 000000000000..7df9f5ae2609
--- /dev/null
+++ b/Documentation/sound/cards/img-spdif-in.rst
@@ -0,0 +1,53 @@
+================================================
+Imagination Technologies SPDIF Input Controllers
+================================================
+
+The Imagination Technologies SPDIF Input controller contains the following
+controls:
+
+* name='IEC958 Capture Mask',index=0
+
+This control returns a mask that shows which of the IEC958 status bits
+can be read using the 'IEC958 Capture Default' control.
+
+* name='IEC958 Capture Default',index=0
+
+This control returns the status bits contained within the SPDIF stream that
+is being received. The 'IEC958 Capture Mask' shows which bits can be read
+from this control.
+
+* name='SPDIF In Multi Frequency Acquire',index=0
+* name='SPDIF In Multi Frequency Acquire',index=1
+* name='SPDIF In Multi Frequency Acquire',index=2
+* name='SPDIF In Multi Frequency Acquire',index=3
+
+This control is used to attempt acquisition of up to four different sample
+rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency'
+control.
+
+When the value of this control is set to {0,0,0,0}, the rate given to hw_params
+will determine the single rate the block will capture. Else, the rate given to
+hw_params will be ignored, and the block will attempt capture for each of the
+four sample rates set here.
+
+If less than four rates are required, the same rate can be specified more than
+once
+
+* name='SPDIF In Lock Frequency',index=0
+
+This control returns the active capture rate, or 0 if a lock has not been
+acquired
+
+* name='SPDIF In Lock TRK',index=0
+
+This control is used to modify the locking/jitter rejection characteristics
+of the block. Larger values increase the locking range, but reduce jitter
+rejection.
+
+* name='SPDIF In Lock Acquire Threshold',index=0
+
+This control is used to change the threshold at which a lock is acquired.
+
+* name='SPDIF In Lock Release Threshold',index=0
+
+This control is used to change the threshold at which a lock is released.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
new file mode 100644
index 000000000000..c016f8c3b88b
--- /dev/null
+++ b/Documentation/sound/cards/index.rst
@@ -0,0 +1,19 @@
+Card-Specific Information
+=========================
+
+.. toctree::
+   :maxdepth: 2
+
+   joystick
+   cmipci
+   sb-live-mixer
+   audigy-mixer
+   emu10k1-jack
+   via82xx-mixer
+   audiophile-usb
+   mixart
+   bt87x
+   maya44
+   hdspm
+   serial-u16550
+   img-spdif-in
diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst
new file mode 100644
index 000000000000..a6e468c81d02
--- /dev/null
+++ b/Documentation/sound/cards/joystick.rst
@@ -0,0 +1,91 @@
+=======================================
+Analog Joystick Support on ALSA Drivers
+=======================================
+
+Oct. 14, 2003
+
+Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+First of all, you need to enable GAMEPORT support on Linux kernel for
+using a joystick with the ALSA driver.  For the details of gameport
+support, refer to Documentation/input/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 [#f1]_	joystick	0 = disable (default), 1 = enable
+ymfpci		joystick_port	0 = disable (default), 1 = auto-detect,
+	                        manual: 0x201, 0x202, 0x204, 0x205 [#f2]_
+==============	=============	============================================
+
+.. [#f1] VIA686A/B only
+.. [#f2] With YMF744/754 chips, the port address can be chosen arbitrarily
+
+The following drivers don't support gameport natively, but there are
+additional modules.  Load the corresponding module to add the gameport
+support.
+
+=======	=================
+Driver	Additional Module
+=======	=================
+emu10k1	emu10k1-gp
+fm801	fm801-gp
+=======	=================
+
+Note: the "pcigame" and "cs461x" modules are for the OSS drivers only.
+These ALSA drivers (cs46xx, trident and au88x0) have the
+built-in gameport support.
+
+As mentioned above, ALSA PCI drivers have the built-in gameport
+support, so you don't have to load ns558 module.  Just load "joydev"
+and the appropriate adapter module (e.g. "analog").
+
+
+ISA Cards
+---------
+
+ALSA ISA drivers don't have the built-in gameport support.
+Instead, you need to load "ns558" module in addition to "joydev" and
+the adapter module (e.g. "analog").
diff --git a/Documentation/sound/cards/maya44.rst b/Documentation/sound/cards/maya44.rst
new file mode 100644
index 000000000000..bf09a584b443
--- /dev/null
+++ b/Documentation/sound/cards/maya44.rst
@@ -0,0 +1,186 @@
+=================================
+Notes on Maya44 USB Audio Support
+=================================
+
+.. note::
+   The following is the original document of Rainer's patch that the
+   current maya44 code based on.  Some contents might be obsoleted, but I
+   keep here as reference -- tiwai
+
+Feb 14, 2008
+
+Rainer Zimmermann <mail@lightshed.de>
+ 
+STATE OF DEVELOPMENT
+====================
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS
+==============
+
+the following files were added:
+
+* pci/ice1724/maya44.c - Maya44 specific code
+* pci/ice1724/maya44.h
+* pci/ice1724/ice1724.patch
+* pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+* i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs 
+* include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+* wtm.h
+* vt1720_mobo.h
+* revo.h
+* prodigy192.h
+* pontis.h
+* phase.h
+* maya44.h
+* juli.h
+* aureon.h
+* amp.h
+* envy24ht.h
+* se.h
+* prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES
+==============
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES
+=============
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+* hw:0,0 input - stereo, analog input 1+2
+* hw:0,0 output - stereo, analog output 1+2
+* hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+* hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS
+========================
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM
+    (digital) output level for channel 1+2
+PCM 1
+    same for channel 3+4
+
+Mic Phantom+48V
+    switch for +48V phantom power for electrostatic microphones on input 1/2.
+
+    Make sure this is not turned on while any other source is connected to input 1/2.
+    It might damage the source and/or the maya44 card.
+
+Mic/Line input
+    if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass
+    analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1
+    same for channel 3+4.
+
+Crossmix
+    cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1
+    cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output
+    switch for S/PDIF output.
+
+    This is not supported by the ESI windows driver.
+    S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors
+    These switches allow a direct digital routing from the ADCs to the DACs.
+    Each switch determines where the digital input data to one of the DACs comes from.
+    They are not supported by the ESI windows driver.
+    For normal operation, they should all be set to "PCM out".
+
+H/W
+    Output source channel 1
+H/W 1
+    Output source channel 2
+H/W 2
+    Output source channel 3
+H/W 3
+    Output source channel 4
+
+H/W 4 ... H/W 9
+    unknown function, left in to enable testing.
+
+    Possibly some of these control S/PDIF output(s).
+    If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+
+PCM out
+	DAC output of the corresponding channel (default setting)
+Input 1 ... Input 4
+	direct routing from ADC output of the selected input channel
+
diff --git a/Documentation/sound/cards/mixart.rst b/Documentation/sound/cards/mixart.rst
new file mode 100644
index 000000000000..48aba98b088f
--- /dev/null
+++ b/Documentation/sound/cards/mixart.rst
@@ -0,0 +1,110 @@
+==============================================================
+Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
+==============================================================
+
+Digigram <alsa@digigram.com>
+
+
+GENERAL
+=======
+
+The miXart8 is a multichannel audio processing and mixing soundcard
+that has 4 stereo audio inputs and 4 stereo audio outputs.
+The miXart8AES/EBU is the same with a add-on card that offers further
+4 digital stereo audio inputs and outputs.
+Furthermore the add-on card offers external clock synchronisation
+(AES/EBU, Word Clock, Time Code and Video Synchro)
+
+The mainboard has a PowerPC that offers onboard mpeg encoding and
+decoding, samplerate conversions and various effects.
+
+The driver don't work properly at all until the certain firmwares
+are loaded, i.e. no PCM nor mixer devices will appear.
+Use the mixartloader that can be found in the alsa-tools package.
+
+
+VERSION 0.1.0
+=============
+
+One miXart8 board will be represented as 4 alsa cards, each with 1
+stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device.
+With a miXart8AES/EBU there is in addition 1 stereo digital input
+'pcm1c' and 1 stereo digital output 'pcm1p' per card.
+
+Formats
+-------
+U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
+Sample rates : 8000 - 48000 Hz continuously
+
+Playback
+--------
+For instance the playback devices are configured to have max. 4
+substreams performing hardware mixing. This could be changed to a
+maximum of 24 substreams if wished.
+Mono files will be played on the left and right channel. Each channel
+can be muted for each stream to use 8 analog/digital outputs separately.
+
+Capture
+-------
+There is one substream per capture device. For instance only stereo
+formats are supported.
+
+Mixer
+-----
+<Master> and <Master Capture>
+	analog volume control of playback and capture PCM.
+<PCM 0-3> and <PCM Capture>
+	digital volume control of each analog substream.
+<AES 0-3> and <AES Capture>
+	digital volume control of each AES/EBU substream.
+<Monitoring>
+	Loopback from 'pcm0c' to 'pcm0p' with digital volume
+	and mute control.
+
+Rem : for best audio quality try to keep a 0 attenuation on the PCM
+and AES volume controls which is set by 219 in the range from 0 to 255
+(about 86% with alsamixer)
+
+
+NOT YET IMPLEMENTED
+===================
+
+- external clock support (AES/EBU, Word Clock, Time Code, Video Sync)
+- MPEG audio formats
+- mono record
+- on-board effects and samplerate conversions
+- linked streams
+
+
+FIRMWARE
+========
+
+[As of 2.6.11, the firmware can be loaded automatically with hotplug
+ when CONFIG_FW_LOADER is set.  The mixartloader is necessary only
+ for older versions or when you build the driver into kernel.]
+ 
+For loading the firmware automatically after the module is loaded, use a
+install command.  For example, add the following entry to
+/etc/modprobe.d/mixart.conf for miXart driver:
+::
+
+	install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
+			   /usr/bin/mixartloader
+
+
+(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to
+/etc/modules.conf, instead.)
+
+The firmware binaries are installed on /usr/share/alsa/firmware
+(or /usr/local/share/alsa/firmware, depending to the prefix option of
+configure).  There will be a miXart.conf file, which define the dsp image
+files.
+
+The firmware files are copyright by Digigram SA
+
+
+COPYRIGHT
+=========
+
+Copyright (c) 2003 Digigram SA <alsa@digigram.com>
+Distributable under GPL.
diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst
new file mode 100644
index 000000000000..bcb62fc99bbb
--- /dev/null
+++ b/Documentation/sound/cards/sb-live-mixer.rst
@@ -0,0 +1,373 @@
+===========================================
+Sound Blaster Live mixer / default DSP code
+===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+IEC958 (S/PDIF) raw PCM
+=======================
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time. 
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+Digital mixer controls
+======================
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC
+	digital to analog converter
+ADC
+	analog to digital converter
+I2S
+	one-way three wire serial bus for digital sound by Philips Semiconductors
+        (this standard is used for connecting standalone DAC and ADC converters)
+LFE
+	low frequency effects (subwoofer signal)
+AC97
+	a chip containing an analog mixer, DAC and ADC converters
+IEC958
+	S/PDIF
+FX-bus
+	the EMU10K1 chip has an effect bus containing 16 accumulators.
+	Each of the synthesizer voices can feed its output to these accumulators
+	and the DSP microcontroller can operate with the resulting sum.
+
+
+``name='Wave Playback Volume',index=0``
+---------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Wave Surround Playback Volume',index=0``
+------------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+``name='Wave Center Playback Volume',index=0``
+----------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+``name='Wave LFE Playback Volume',index=0``
+-------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+``name='Wave Capture Volume',index=0``, ``name='Wave Capture Switch',index=0``
+------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Synth Playback Volume',index=0``
+----------------------------------------
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Synth Capture Volume',index=0``, ``name='Synth Capture Switch',index=0``
+--------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Surround Playback Volume',index=0``
+-------------------------------------------
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+``name='Surround Capture Volume',index=0``, ``name='Surround Capture Switch',index=0``
+--------------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Center Playback Volume',index=0``
+-----------------------------------------
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+``name='LFE Playback Volume',index=0``
+--------------------------------------
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+``name='AC97 Playback Volume',index=0``
+---------------------------------------
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+
+.. note::
+  This control should be zero for the standard operations, otherwise
+  a digital loopback is activated.
+
+
+``name='AC97 Capture Volume',index=0``
+--------------------------------------
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+
+.. note::
+   This control should be 100 (maximal value), otherwise no analog
+   inputs of the AC97 codec can be captured (recorded).
+
+``name='IEC958 TTL Playback Volume',index=0``
+---------------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='IEC958 TTL Capture Volume',index=0``
+--------------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+``name='Zoom Video Playback Volume',index=0``
+---------------------------------------------
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Zoom Video Capture Volume',index=0``
+--------------------------------------------
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+``name='IEC958 LiveDrive Playback Volume',index=0``
+---------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+``name='IEC958 LiveDrive Capture Volume',index=0``
+--------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+``name='IEC958 Coaxial Playback Volume',index=0``
+-------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+``name='IEC958 Coaxial Capture Volume',index=0``
+------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+``name='Line LiveDrive Playback Volume',index=0``, ``name='Line LiveDrive Playback Volume',index=1``
+----------------------------------------------------------------------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+``name='Line LiveDrive Capture Volume',index=1``, ``name='Line LiveDrive Capture Volume',index=1``
+--------------------------------------------------------------------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+``name='Tone Control - Switch',index=0``
+----------------------------------------
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+``name='Tone Control - Bass',index=0``
+--------------------------------------
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+``name='Tone Control - Treble',index=0``
+----------------------------------------
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+``name='IEC958 Optical Raw Playback Switch',index=0``
+-----------------------------------------------------
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+``name='Headphone Playback Volume',index=1``
+--------------------------------------------
+This control attenuates the samples for the headphone output.
+
+``name='Headphone Center Playback Switch',index=1``
+---------------------------------------------------
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+``name='Headphone LFE Playback Switch',index=1``
+------------------------------------------------
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+PCM stream related controls
+===========================
+
+``name='EMU10K1 PCM Volume',index 0-31``
+----------------------------------------
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+* 0 - mono, default 0xffff (no attenuation)
+* 1 - left, default 0xffff (no attenuation)
+* 2 - right, default 0xffff (no attenuation)
+
+``name='EMU10K1 PCM Send Routing',index 0-31``
+----------------------------------------------
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+*  0 -  mono, A destination (FX-bus 0-15), default 0
+*  1 -  mono, B destination (FX-bus 0-15), default 1
+*  2 -  mono, C destination (FX-bus 0-15), default 2
+*  3 -  mono, D destination (FX-bus 0-15), default 3
+*  4 -  left, A destination (FX-bus 0-15), default 0
+*  5 -  left, B destination (FX-bus 0-15), default 1
+*  6 -  left, C destination (FX-bus 0-15), default 2
+*  7 -  left, D destination (FX-bus 0-15), default 3
+*  8 - right, A destination (FX-bus 0-15), default 0
+*  9 - right, B destination (FX-bus 0-15), default 1
+* 10 - right, C destination (FX-bus 0-15), default 2
+* 11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+``name='EMU10K1 PCM Send Volume',index 0-31``
+---------------------------------------------
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+*  0 -  mono, A destination attn, default 255 (no attenuation)
+*  1 -  mono, B destination attn, default 255 (no attenuation)
+*  2 -  mono, C destination attn, default 0 (mute)
+*  3 -  mono, D destination attn, default 0 (mute)
+*  4 -  left, A destination attn, default 255 (no attenuation)
+*  5 -  left, B destination attn, default 0 (mute)
+*  6 -  left, C destination attn, default 0 (mute)
+*  7 -  left, D destination attn, default 0 (mute)
+*  8 - right, A destination attn, default 0 (mute)
+*  9 - right, B destination attn, default 255 (no attenuation)
+* 10 - right, C destination attn, default 0 (mute)
+* 11 - right, D destination attn, default 0 (mute)
+
+
+
+MANUALS/PATENTS
+===============
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+LM4545.pdf
+	AC97 Codec
+m2049.pdf
+	The EMU10K1 Digital Audio Processor
+hog63.ps
+	FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+
+WO 9901813 (A1)
+	Audio Effects Processor with multiple asynchronous streams
+	(Jan. 14, 1999)
+
+WO 9901814 (A1)
+	Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+WO 9901953 (A1)
+	Audio Effects Processor having Decoupled Instruction
+        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (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
+        with a multiport memory onto which multiple asynchronous
+        digital sound samples can be concurrently loaded
+	(Jul. 27, 1999)
+
+US 5930158
+	Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+US 6032235
+	Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+US 6138207
+	Interpolation looping of audio samples in cache connected to
+        system bus with prioritization and modification of bus transfers
+        in accordance with loop ends and minimum block sizes
+	(Oct. 24, 2000)
+
+US 6151670
+	Method for conserving memory storage using a
+        pool of  short term memory registers
+	(Nov. 21, 2000)
+
+US 6195715
+	Interrupt control for multiple programs communicating with
+        a common interrupt by associating programs to GP registers,
+        defining interrupt register, polling GP registers, and invoking
+        callback routine associated with defined interrupt register
+	(Feb. 27, 2001)
diff --git a/Documentation/sound/cards/serial-u16550.rst b/Documentation/sound/cards/serial-u16550.rst
new file mode 100644
index 000000000000..197aeacea3da
--- /dev/null
+++ b/Documentation/sound/cards/serial-u16550.rst
@@ -0,0 +1,93 @@
+===================================
+Serial UART 16450/16550 MIDI driver
+===================================
+
+The adaptor module parameter allows you to select either:
+
+* 0 - Roland Soundcanvas support (default)
+* 1 - Midiator MS-124T support (1)
+* 2 - Midiator MS-124W S/A mode (2)
+* 3 - MS-124W M/B mode support (3)
+* 4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15).  Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1.  Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors.  Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+			speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16.  The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4.  Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports.  M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs. 
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported. 
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with.  The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port.  Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow.  The number of
+inputs is specified by the ins parameter.
diff --git a/Documentation/sound/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst
new file mode 100644
index 000000000000..6ee993d4535b
--- /dev/null
+++ b/Documentation/sound/cards/via82xx-mixer.rst
@@ -0,0 +1,8 @@
+=============
+VIA82xx mixer
+=============
+
+On many VIA82xx boards, the ``Input Source Select`` mixer control does not work.
+Setting it to ``Input2`` on such boards will cause recording to hang, or fail
+with EIO (input/output error) via OSS emulation.  This control should be left
+at ``Input1`` for such cards.