Merge branch 'master' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6

8 files changed, 246 insertions, 841 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 355ff0a2bb7c..241e26c4ff92 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     above explicitly.
 
     The power-management is supported.
-    
+
+  Module snd-cs5530
+  _________________
+
+    Module for Cyrix/NatSemi Geode 5530 chip. 
+  
   Module snd-cs5535audio
   ----------------------
 
@@ -759,6 +764,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     model	- force the model name
     position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
+    probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
     single_cmd  - Use single immediate commands to communicate with
 		codecs (for debugging only)
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
@@ -803,6 +809,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  hp-3013	HP machines (3013-variant)
 	  fujitsu	Fujitsu S7020
 	  acer		Acer TravelMate
+	  will		Will laptops (PB V7900)
+	  replacer	Replacer 672V
 	  basic		fixed pin assignment (old default model)
 	  auto		auto-config reading BIOS (default)
 
@@ -811,16 +819,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  hp-bpc	HP xw4400/6400/8400/9400 laptops
 	  hp-bpc-d7000	HP BPC D7000
 	  benq		Benq ED8
+	  benq-t31	Benq T31
 	  hippo		Hippo (ATI) with jack detection, Sony UX-90s
 	  hippo_1	Hippo (Benq) with jack detection
+	  sony-assamd	Sony ASSAMD
 	  basic		fixed pin assignment w/o SPDIF
 	  auto		auto-config reading BIOS (default)
 
+	ALC268
+	  3stack	3-stack model
+	  auto		auto-config reading BIOS (default)
+
+	ALC662
+	  3stack-dig	3-stack (2-channel) with SPDIF
+	  3stack-6ch	 3-stack (6-channel)
+	  3stack-6ch-dig 3-stack (6-channel) with SPDIF
+	  6stack-dig	 6-stack with SPDIF
+	  lenovo-101e	 Lenovo laptop
+	  auto		auto-config reading BIOS (default)
+
 	ALC882/885
 	  3stack-dig	3-jack with SPDIF I/O
 	  6stack-dig	6-jack digital with SPDIF I/O
 	  arima		Arima W820Di1
 	  macpro	MacPro support
+	  imac24	iMac 24'' with jack detection
 	  w2jc		ASUS W2JC
 	  auto		auto-config reading BIOS (default)
 
@@ -832,9 +855,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  6stack-dig-demo  6-jack digital for Intel demo board
 	  acer		Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
 	  medion	Medion Laptops
+	  medion-md2	Medion MD2
 	  targa-dig	Targa/MSI
 	  targa-2ch-dig	Targs/MSI with 2-channel
 	  laptop-eapd   3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
+	  lenovo-101e	Lenovo 101E
+	  lenovo-nb0763	Lenovo NB0763
+	  lenovo-ms7195-dig Lenovo MS7195
+	  6stack-hp	HP machines with 6stack (Nettle boards)
+	  3stack-hp	HP machines with 3stack (Lucknow, Samba boards)
 	  auto		auto-config reading BIOS (default)
 
 	ALC861/660
@@ -853,7 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack-dig	3-jack with SPDIF OUT
 	  6stack-dig	6-jack with SPDIF OUT
 	  3stack-660	3-jack (for ALC660VD)
+	  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
 	  lenovo	Lenovo 3000 C200
+	  dallas	Dallas laptops
 	  auto		auto-config reading BIOS (default)
 
 	CMI9880
@@ -864,12 +895,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  allout	5-jack in back, 2-jack in front, SPDIF out
 	  auto		auto-config reading BIOS (default)
 
+	AD1882
+	  3stack	3-stack mode (default)
+	  6stack	6-stack mode
+
+	AD1884
+	  N/A
+
 	AD1981
 	  basic		3-jack (default)
 	  hp		HP nx6320
 	  thinkpad	Lenovo Thinkpad T60/X60/Z60
 	  toshiba	Toshiba U205
 
+	AD1983
+	  N/A
+
+	AD1984
+	  basic		default configuration
+	  thinkpad	Lenovo Thinkpad T61/X61
+
 	AD1986A
 	  6stack	6-jack, separate surrounds (default)
 	  3stack	3-stack, shared surrounds
@@ -907,11 +952,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  ref		Reference board
 	  3stack	D945 3stack
 	  5stack	D945 5stack + SPDIF
-	  macmini	Intel Mac Mini
-	  macbook	Intel Mac Book
-	  macbook-pro-v1 Intel Mac Book Pro 1st generation
-	  macbook-pro	Intel Mac Book Pro 2nd generation
-	  imac-intel	Intel iMac
+	  dell		Dell XPS M1210
+	  intel-mac-v1	Intel Mac Type 1
+	  intel-mac-v2	Intel Mac Type 2
+	  intel-mac-v3	Intel Mac Type 3
+	  intel-mac-v4	Intel Mac Type 4
+	  intel-mac-v5	Intel Mac Type 5
+	  macmini	Intel Mac Mini (equivalent with type 3)
+	  macbook	Intel Mac Book (eq. type 5)
+	  macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+	  macbook-pro	Intel Mac Book Pro 2nd generation (eq. type 3)
+	  imac-intel	Intel iMac (eq. type 2)
+	  imac-intel-20	Intel iMac (newer version) (eq. type 3)
 
 	STAC9202/9250/9251
 	  ref		Reference board, base config
@@ -956,6 +1008,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     from the irq.  Remember this is a last resort, and should be
     avoided as much as possible...
     
+    MORE NOTES ON "azx_get_response timeout" PROBLEMS:
+    On some hardwares, you may need to add a proper probe_mask option
+    to avoid the "azx_get_response timeout" problem above, instead.
+    This occurs when the access to non-existing or non-working codec slot
+    (likely a modem one) causes a stall of the communication via HD-audio
+    bus.  You can see which codec slots are probed by enabling
+    CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec
+    proc files.  Then limit the slots to probe by probe_mask option.
+    For example, probe_mask=1 means to probe only the first slot, and
+    probe_mask=4 means only the third slot.
+
     The power-management is supported.
 
   Module snd-hdsp
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index e40cce83327c..2ad5e6306c44 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -1,4 +1,4 @@
-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.3
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.5
 	========================================================
 
 	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -6,8 +6,19 @@
 This document is a guide to using the M-Audio Audiophile USB (tm) device with 
 ALSA and JACK.
 
+History
+=======
+* v1.4 - Thibault Le Meur (2007-07-11)
+ - Added Low Endianness nature of 16bits-modes
+   found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+ - Modifying document structure
+* v1.5 - Thibault Le Meur (2007-07-12)
+ - Added AC3/DTS passthru info
+
+
 1 - Audiophile USB Specs and correct usage
 ==========================================
+
 This part is a reminder of important facts about the functions and limitations 
 of the device.
 
@@ -25,18 +36,18 @@ The device has 4 audio interfaces, and 2 MIDI ports:
 The internal DAC/ADC has the following characteristics:
 * sample depth of 16 or 24 bits
 * sample rate from 8kHz to 96kHz
-* Two ports 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"
+* 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
+ * 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
+ * 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)
+ * 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:
@@ -52,44 +63,56 @@ source is connected
 synchronization error (for instance sound played at an odd sample rate)
 
 
-2 - Audiophile USB support in ALSA
-==================================
+2 - Audiophile USB MIDI support in ALSA
+=======================================
 
-2.1 - MIDI ports
-----------------
-The Audiophile USB MIDI ports will be automatically supported once the 
+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.
 
-2.2 - Audio ports
------------------
+
+3 - Audiophile USB Audio support in ALSA
+========================================
 
 Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
 module. This module can work in a default mode (without any device-specific 
 parameter), or in an "advanced" mode with the device-specific parameter called 
 "device_setup".
 
-2.2.1 - Default Alsa driver mode
-
-The default behavior of the snd-usb-audio driver is to parse the device 
-capabilities at startup and enable all functions inside the device (including 
-all ports at any supported sample rates and sample depths). This approach 
-has the advantage to let the driver easily switch from sample rates/depths 
-automatically according to the need of the application claiming the device.
-
-In this case the Audiophile ports are mapped to alsa pcm devices in the 
-following way (I suppose the device's index is 1):
+3.1 - Default Alsa driver mode
+------------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device 
+capabilities at startup and activate the required mode when required 
+by the applications: for instance if the user is recording in a 
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample 
+rates/depths automatically according to the user's needs. However, those who 
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this 
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now.  In this case the 
+Audiophile interfaces are mapped to alsa pcm devices in the following 
+way (I suppose the device's index is 1):
  * hw:1,0 is Ao in playback and Di in capture
  * hw:1,1 is Do in playback and Ai in capture
  * hw:1,2 is Do in AC3/DTS passthrough mode
 
-You must note as well that 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 is Little Endian 
-compliant and thus uses S16_LE.
+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
@@ -98,22 +121,26 @@ Examples:
    % 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 setup and don't experience any 
+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.
 
-2.2.2 - Advanced module setup
+3.2 - Advanced module setup
+---------------------------
 
 Due to the hardware constraints described above, the device initialization made 
 by the Alsa driver in default mode may result in a corrupted state of the 
 device. For instance, a particularly annoying issue is that the sound captured 
-from the Ai port sounds distorted (as if boosted with an excessive high volume 
-gain).
+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".
+parameter called "device_setup" (this parameter was introduced in kernel
+release 2.6.17)
 
-2.2.2.1 - Initializing the working mode of the Audiophile USB
+3.2.1 - Initializing the working mode of the Audiophile USB
 
 As far as the Audiophile USB device is concerned, this value let the user 
 specify:
@@ -121,33 +148,57 @@ specify:
  * the sample rate
  * whether the Di port is used or not 
 
-Here is a list of supported device_setup values for this device:
- * device_setup=0x00 (or omitted)
-   - Alsa driver default mode
-   - maintains backward compatibility with setups that do not use this 
-     parameter by not introducing any change
-   - results sometimes in corrupted sound as described earlier
+When initialized with "device_setup=0x00", the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default 
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+3.2.1.1 - 16-bit modes
+
+The two supported modes are:
+
  * device_setup=0x01
    - 16bits 48kHz mode with Di disabled
    - Ai,Ao,Do can be used at the same time
    - hw:1,0 is not available in capture mode
    - hw:1,2 is not available
+
  * device_setup=0x11
    - 16bits 48kHz mode with Di enabled
    - Ai,Ao,Di,Do can be used at the same time
    - hw:1,0 is available in capture mode
    - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+3.2.1.2 - 24-bit modes
+
+The three supported modes are:
+
  * device_setup=0x09
    - 24bits 48kHz mode with Di disabled
    - Ai,Ao,Do can be used at the same time
    - hw:1,0 is not available in capture mode
    - hw:1,2 is not available
+
  * device_setup=0x19
    - 24bits 48kHz mode with Di enabled
    - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
    - hw:1,0 is available in capture mode and an active digital source must be 
      connected to Di
    - hw:1,2 is not available
+
  * device_setup=0x0D or 0x10
    - 24bits 96kHz mode
    - Di is enabled by default for this mode but does not need to be connected 
@@ -155,34 +206,64 @@ Here is a list of supported device_setup values for this device:
    - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
    - hw:1,0 is available in captured mode
    - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
+mode" above for an aplay command example)
+
+3.2.1.3 - AC3 w/ DTS passthru mode
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
  * device_setup=0x03
    - 16bits 48kHz mode with only the Do port enabled 
-   - AC3 with DTS passthru (not tested)
+   - AC3 with DTS passthru
    - Caution with this setup the Do port is mapped to the pcm device hw:1,0
 
-2.2.2.2 - Setting and switching configurations with the device_setup parameter
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+3.2.2 - How to use the device_setup parameter
+----------------------------------------------
 
 The parameter can be given:
+
  * By manually probing the device (as root):
    # modprobe -r snd-usb-audio
    # modprobe snd-usb-audio index=1 device_setup=0x09
+
  * Or while configuring the modules options in your modules configuration file
    - For Fedora distributions, edit the /etc/modprobe.conf file:
        alias snd-card-1 snd-usb-audio
        options snd-usb-audio index=1 device_setup=0x09
 
-IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
--------------------------------------------
- * You may need to _first_ initialize the module with the correct device_setup 
-   parameter and _only_after_ turn on the Audiophile USB device
- * This is especially true when switching the sample depth:
+CAUTION when initializaing 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 in a misconfiguration of the device. In this case
+   turn off the device, unproble the snd-usb-audio module, then probe it again with 
+   correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+   to  another mode (possibly with another sample-depth), please use also the following 
+   procedure:
    - first turn off the device
    - de-register the snd-usb-audio module (modprobe -r)
    - change the device_setup parameter by changing the device_setup
      option in /etc/modprobe.conf 
    - turn on the device
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+   be enough to ensure the 'stability' of the device initialization.
 
-2.2.2.3 - Audiophile USB's device_setup structure
+3.2.3 - Technical details for hackers
+-------------------------------------
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+3.2.3.1 - Audiophile USB's device_setup structure
 
 If you want to understand the device_setup magic numbers for the Audiophile 
 USB, you need some very basic understanding of binary computation. However, 
@@ -228,12 +309,12 @@ Caution:
    - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
      only be able to use one at the same time
 
-2.2.3 -  USB implementation details for this device
+3.2.3.2 -  USB implementation details for this device
 
 You may safely skip this section if you're not interested in driver 
-development.
+hacking.
 
-This section describes some internal aspects of the device and summarize the 
+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:
@@ -293,43 +374,45 @@ 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.
 
-3 - Audiophile USB and Jack support
+4 - Audiophile USB and Jack support
 ===================================
 
 This section deals with support of the Audiophile USB device in Jack.
-The main issue regarding this support is that the device is Big Endian 
-compliant.
 
-3.1 - Using the plug alsa plugin
---------------------------------
+There are 2 main potential issues when using Jackd with the device:
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+4.1 - Direct support in Jackd
+-----------------------------
 
-Jack doesn't directly support big endian devices. Thus, one way to have support 
-for this device with Alsa is to use the Alsa "plug" converter.
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember 
+extacly when this support was released into jackd, let's just say that 
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
+16bits Big-Endian devices, but since you've read  carefully the above 
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+4.2 - Using Alsa plughw
+-----------------------
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa "plug" converter.
 
 For instance here is one way to run Jack with 2 playback channels on Ao and 2 
 capture channels from Ai:
   % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
 
-
 However you may see the following warning message:
 "You appear to be using the ALSA software "plug" layer, probably a result of 
 using the "default" ALSA device. This is less efficient than it could be. 
 Consider using a hardware device instead rather than using the plug layer."
 
-3.2 - Patching alsa to use direct pcm device
---------------------------------------------
-A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. 
-However it has not been included in the CVS tree.
-
-You can find it at the following URL:
-http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
-atid=425939
-
-After having applied the patch you can run jackd with the following command 
-line:
-  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
-
-3.2 - Getting 2 input and/or output interfaces in Jack
+4.3 - Getting 2 input and/or output interfaces in Jack
 ------------------------------------------------------
 
 As you can see, starting the Jack server this way will only enable 1 stereo
@@ -339,6 +422,7 @@ 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.
 
@@ -348,13 +432,11 @@ 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:
-* patching Jack with the previously mentioned "Big Endian" patch
-* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
-* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* 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, but this may be due to my OS
-configuration. If you have any success with this kind of setup, please
-drop me an email.
+I had no success in testing this for now, if you have any success with this kind 
+of setup, please drop me an email.
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
index ec2a02541d5b..bfa0c9aacb4b 100644
--- a/Documentation/sound/alsa/OSS-Emulation.txt
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -278,6 +278,21 @@ current mixer configuration by reading and writing the whole file
 image.
 
 
+Duplex Streams
+==============
+
+Note that when attempting to use a single device file for playback and
+capture, the OSS API provides no way to set the format, sample rate or
+number of channels different in each direction.  Thus
+	io_handle = open("device", O_RDWR)
+will only function correctly if the values are the same in each direction.
+
+To use different values in the two directions, use both
+	input_handle = open("device", O_RDONLY)
+	output_handle = open("device", O_WRONLY)
+and set the values for the corresponding handle.
+
+
 Unsupported Features
 ====================
 
diff --git a/Documentation/sound/oss/AD1816 b/Documentation/sound/oss/AD1816
deleted file mode 100644
index 14bd8f25d523..000000000000
--- a/Documentation/sound/oss/AD1816
+++ /dev/null
@@ -1,84 +0,0 @@
-Documentation for the AD1816(A) sound driver
-============================================
-
-Installation:
--------------
-
-To get your AD1816(A) based sound card work, you'll have to enable support for
-experimental code ("Prompt for development and/or incomplete code/drivers")
-and isapnp ("Plug and Play support", "ISA Plug and Play support"). Enable
-"Sound card support", "OSS modules support" and "Support for AD1816(A) based
-cards (EXPERIMENTAL)" in the sound configuration menu, too. Now build, install
-and reboot the new kernel as usual.
-
-Features:
----------
-
-List of features supported by this driver:
-- full-duplex support
-- supported audio formats: unsigned 8bit, signed 16bit little endian, 
-                           signed 16bit big endian, µ-law, A-law
-- supported channels: mono and stereo
-- supported recording sources: Master, CD, Line, Line1, Line2, Mic
-- supports phat 3d stereo circuit (Line 3)
-
-
-Supported cards:
-----------------
-
-The following cards are known to work with this driver:
-- Terratec Base 1 
-- Terratec Base 64 
-- HP Kayak 
-- Acer FX-3D
-- SY-1816 
-- Highscreen Sound-Boostar 32 Wave 3D
-- Highscreen Sound-Boostar 16 
-- AVM Apex Pro card 
-- (Aztech SC-16 3D) 
-- (Newcom SC-16 3D) 
-- (Terratec EWS64S) 
-
-Cards listed in brackets are not supported reliable. If you have such a card 
-you should add the extra parameter:
-  options=1
-when loading the ad1816 module via modprobe. 
-
-
-Troubleshooting:
-----------------
-
-First of all you should check, if the driver has been loaded
-properly.
-
-If loading of the driver succeeds, but playback/capture fails, check
-if you used the correct values for irq, dma and dma2 when loading the module.
-If one of them is wrong you usually get the following error message:
-
-Nov  6 17:06:13 tek01 kernel: Sound: DMA (output) timed out - IRQ/DRQ config error?
-
-If playback/capture is too fast or to slow, you should have a look at
-the clock chip of your sound card. The AD1816 was designed for a 33MHz
-oscillator, however most sound card manufacturer use slightly
-different oscillators as they are cheaper than 33MHz oscillators. If
-you have such a card you have to adjust the ad1816_clockfreq parameter
-above. For example: For a card using a 32.875MHz oscillator use
-ad1816_clockfreq=32875 instead of ad1816_clockfreq=33000.
-
-
-Updates, bugfixes and bugreports:
---------------------------------
-
-As the driver is still experimental and under development, you should
-watch out for updates.  Updates of the driver are available on the
-Internet from one of my home pages:
-  http://www.student.informatik.tu-darmstadt.de/~tek/projects/linux.html
-or:
-  http://www.tu-darmstadt.de/~tek01/projects/linux.html
-
-Bugreports, bugfixes and related questions should be sent via E-Mail to:
-  tek@rbg.informatik.tu-darmstadt.de
-
-Thorsten Knabe <tek@rbg.informatik.tu-darmstadt.de>
-Christoph Hellwig <hch@infradead.org>
-	Last modified: 2000/09/20
diff --git a/Documentation/sound/oss/NM256 b/Documentation/sound/oss/NM256
deleted file mode 100644
index b503217488b3..000000000000
--- a/Documentation/sound/oss/NM256
+++ /dev/null
@@ -1,280 +0,0 @@
-=======================================================
-Documentation for the NeoMagic 256AV/256ZX sound driver
-=======================================================
-
-You're looking at version 1.1 of the driver.  (Woohoo!) It has been
-successfully tested against the following laptop models:
-
-	Sony Z505S/Z505SX/Z505DX/Z505RX
-	Sony F150, F160, F180, F250, F270, F280, PCG-F26
-	Dell Latitude CPi, CPt (various submodels)
-
-There are a few caveats, which is why you should read the entirety of
-this document first.
-
-This driver was developed without any support or assistance from
-NeoMagic.  There is no warranty, expressed, implied, or otherwise.  It
-is free software in the public domain; feel free to use it, sell it,
-give it to your best friends, even claim that you wrote it (but why?!)
-but don't go whining to me, NeoMagic, Sony, Dell, or anyone else
-when it blows up your computer.
-
-Version 1.1 contains a change to try and detect non-AC97 versions of
-the hardware, and not install itself appropriately.  It should also
-reinitialize the hardware on an APM resume event, assuming that APM
-was configured into your kernel.
-
-============
-Installation
-============
-
-Enable the sound drivers, the OSS sound drivers, and then the NM256
-driver.  The NM256 driver *must* be configured as a module (it won't
-give you any other choice).
-
-Next, do the usual "make modules" and "make modules_install".
-Finally, insmod the soundcore, sound and nm256 modules.
-
-When the nm256 driver module is loaded, you should see a couple of
-confirmation messages in the kernel logfile indicating that it found
-the device (the device does *not* use any I/O ports or DMA channels).
-Now try playing a wav file, futz with the CD-ROM if you have one, etc.
-
-The NM256 is entirely a PCI-based device, and all the necessary
-information is automatically obtained from the card.  It can only be
-configured as a module in a vain attempt to prevent people from
-hurting themselves.  It works correctly if it shares an IRQ with
-another device (it normally shares IRQ 9 with the builtin eepro100
-ethernet on the Sony Z505 laptops). 
-
-It does not run the card in any sort of compatibility mode. It will
-not work on laptops that have the SB16-compatible, AD1848-compatible
-or CS4232-compatible codec/mixer; you will want to use the appropriate
-compatible OSS driver with these chipsets.  I cannot provide any
-assistance with machines using the SB16, AD1848 or CS4232 compatible
-versions.  (The driver now attempts to detect the mixer version, and
-will refuse to load if it believes the hardware is not
-AC97-compatible.)
-
-The sound support is very basic, but it does include simultaneous
-playback and record capability.  The mixer support is also quite
-simple, although this is in keeping with the rather limited
-functionality of the chipset.
-
-There is no hardware synthesizer available, as the Losedows OPL-3 and
-MIDI support is done via hardware emulation.
-
-Only three recording devices are available on the Sony: the
-microphone, the CD-ROM input, and the volume device (which corresponds
-to the stereo output).  (Other devices may be available on other
-models of laptops.)  The Z505 series does not have a builtin CD-ROM,
-so of course the CD-ROM input doesn't work.  It does work on laptops
-with a builtin CD-ROM drive.
-
-The mixer device does not appear to have any tone controls, at least
-on the Z505 series.  The mixer module checks for tone controls in the
-AC97 mixer, and will enable them if they are available.
-
-==============
-Known problems
-==============
-
-  * There are known problems with PCMCIA cards and the eepro100 ethernet 
-    driver on the Z505S/Z505SX/Z505DX.  Keep reading.
-
-  * There are also potential problems with using a virtual X display, and
-    also problems loading the module after the X server has been started. 
-    Keep reading.
-
-  * The volume control isn't anywhere near linear.  Sorry.  This will be
-    fixed eventually, when I get sufficiently annoyed with it.  (I doubt
-    it will ever be fixed now, since I've never gotten sufficiently
-    annoyed with it and nobody else seems to care.)
-
-  * There are reports that the CD-ROM volume is very low.  Since I do not
-    have a CD-ROM equipped laptop, I cannot test this (it's kinda hard to
-    do remotely).
-
-  * Only 8 fixed-rate speeds are supported.  This is mainly a chipset
-    limitation.  It may be possible to support other speeds in the future.
-
-  * There is no support for the telephone mixer/codec.  There is support
-    for a phonein/phoneout device in the mixer driver;  whether or not 
-    it does anything is anyone's guess.  (Reports on this would be
-    appreciated.  You'll have to figure out how to get the phone to
-    go off-hook before it'll work, tho.)
-
-  * This driver was not written with any cooperation or support from
-    NeoMagic.  If you have any questions about this, see their website
-    for their official stance on supporting open source drivers.
-
-============
-Video memory
-============
-
-The NeoMagic sound engine uses a portion of the display memory to hold
-the sound buffer.  (Crazy, eh?)  The NeoMagic video BIOS sets up a
-special pointer at the top of video RAM to indicate where the top of
-the audio buffer should be placed.
-
-At the present time XFree86 is apparently not aware of this.  It will
-thus write over either the pointer or the sound buffer with abandon.
-(Accelerated-X seems to do a better job here.)
-
-This implies a few things:
-
-  * Sometimes the NM256 driver has to guess at where the buffer 
-    should be placed, especially if the module is loaded after the
-    X server is started.  It's usually correct, but it will consistently
-    fail on the Sony F250.
-
-  * Virtual screens greater than 1024x768x16 under XFree86 are
-    problematic on laptops with only 2.5MB of screen RAM. This
-    includes all of the 256AV-equipped laptops.  (Virtual displays
-    may or may not work on the 256ZX, which has at least 4MB of
-    video RAM.)
-
-If you start having problems with random noise being output either
-constantly (this is the usual symptom on the F250), or when windows
-are moved around (this is the usual symptom when using a virtual
-screen), the best fix is to
-
-  * Don't use a virtual frame buffer.
-  * Make sure you load the NM256 module before the X server is
-    started.
-
-On the F250, it is possible to force the driver to load properly even
-after the XFree86 server is started by doing:
-
-	insmod nm256 buffertop=0x25a800
-
-This forces the audio buffers to the correct offset in screen RAM.
-
-One user has reported a similar problem on the Sony F270, although
-others apparently aren't seeing any problems.  His suggested command
-is
-
-	insmod nm256 buffertop=0x272800
-
-=================
-Official WWW site
-=================
-
-The official site for the NM256 driver is:
-
-	http://www.uglx.org/sony.html
-
-You should always be able to get the latest version of the driver there,
-and the driver will be supported for the foreseeable future.
-
-==============
-Z505RX and IDE
-==============
-
-There appears to be a problem with the IDE chipset on the Z505RX; one
-of the symptoms is that sound playback periodically hangs (when the
-disk is accessed).  The user reporting the problem also reported that
-enabling all of the IDE chipset workarounds in the kernel solved the
-problem, tho obviously only one of them should be needed--if someone
-can give me more details I would appreciate it.
-
-==============================
-Z505S/Z505SX on-board Ethernet
-==============================
-
-If you're using the on-board Ethernet Pro/100 ethernet support on the Z505
-series, I strongly encourage you to download the latest eepro100 driver from
-Donald Becker's site:
-
-	ftp://cesdis.gsfc.nasa.gov/pub/linux/drivers/test/eepro100.c
-
-There was a reported problem on the Z505SX that if the ethernet
-interface is disabled and reenabled while the sound driver is loaded,
-the machine would lock up.  I have included a workaround that is
-working satisfactorily.  However, you may occasionally see a message
-about "Releasing interrupts, over 1000 bad interrupts" which indicates
-that the workaround is doing its job.
-
-==================================
-PCMCIA and the Z505S/Z505SX/Z505DX
-==================================
-
-There is also a known problem with the Sony Z505S and Z505SX hanging
-if a PCMCIA card is inserted while the ethernet driver is loaded, or
-in some cases if the laptop is suspended.  This is caused by tons of
-spurious IRQ 9s, probably generated from the PCMCIA or ACPI bridges.
-
-There is currently no fix for the problem that works in every case.
-The only known workarounds are to disable the ethernet interface
-before inserting or removing a PCMCIA card, or with some cards
-disabling the PCMCIA card before ejecting it will also help the
-problem with the laptop hanging when the card is ejected.
-
-One user has reported that setting the tcic's cs_irq to some value
-other than 9 (like 11) fixed the problem.  This doesn't work on my
-Z505S, however--changing the value causes the cardmgr to stop seeing
-card insertions and removals, cards don't seem to work correctly, and
-I still get hangs if a card is inserted when the kernel is booted.
-
-Using the latest ethernet driver and pcmcia package allows me to
-insert an Adaptec 1480A SlimScsi card without the laptop hanging,
-although I still have to shut down the card before ejecting or
-powering down the laptop.  However, similar experiments with a DE-660
-ethernet card still result in hangs when the card is inserted.  I am
-beginning to think that the interrupts are CardBus-related, since the
-Adaptec card is a CardBus card, and the DE-660 is not; however, I
-don't have any other CardBus cards to test with.
-
-======
-Thanks
-======
-
-First, I want to thank everyone (except NeoMagic of course) for their
-generous support and encouragement.  I'd like to list everyone's name
-here that replied during the development phase, but the list is
-amazingly long.
-
-I will be rather unfair and single out a few people, however:
-
-	Justin Maurer, for being the first random net.person to try it,
-	and for letting me login to his Z505SX to get it working there
-
-	Edi Weitz for trying out several different versions, and giving
-	me a lot of useful feedback
-
-	Greg Rumple for letting me login remotely to get the driver 
-	functional on the 256ZX, for his assistance on tracking
-	down all sorts of random stuff, and for trying out Accel-X
-
-	Zach Brown, for the initial AC97 mixer interface design
-
-	Jeff Garzik, for various helpful suggestions on the AC97
-	interface
-
-	"Mr. Bumpy" for feedback on the Z505RX
-
-	Bill Nottingham, for generous assistance in getting the mixer ID
-	code working
-
-=================
-Previous versions
-=================
-
-Versions prior to 0.3 (aka `noname') had problems with weird artifacts
-in the output and failed to set the recording rate properly.  These
-problems have long since been fixed.
-
-Versions prior to 0.5 had problems with clicks in the output when
-anything other than 16-bit stereo sound was being played, and also had
-periodic clicks when recording.
-
-Version 0.7 first incorporated support for the NM256ZX chipset, which
-is found on some Dell Latitude laptops (the CPt, and apparently
-some CPi models as well).  It also included the generic AC97
-mixer module.
-
-Version 0.75 renamed all the functions and files with slightly more
-generic names.
-
-Note that previous versions of this document claimed that recording was
-8-bit only; it actually has been working for 16-bits all along.
diff --git a/Documentation/sound/oss/OPL3-SA2 b/Documentation/sound/oss/OPL3-SA2
deleted file mode 100644
index d8b6d2bbada6..000000000000
--- a/Documentation/sound/oss/OPL3-SA2
+++ /dev/null
@@ -1,210 +0,0 @@
-Documentation for the OPL3-SA2, SA3, and SAx driver (opl3sa2.o)
----------------------------------------------------------------
-
-Scott Murray, scott@spiteful.org
-January 7, 2001
-
-NOTE: All trade-marked terms mentioned below are properties of their
-      respective owners.
-
-
-Supported Devices
------------------
-
-This driver is for PnP soundcards based on the following Yamaha audio
-controller chipsets:
-
-YMF711 aka OPL3-SA2
-YMF715 and YMF719 aka OPL3-SA3
-
-Up until recently (December 2000), I'd thought the 719 to be a
-different chipset, the OPL3-SAx.  After an email exhange with
-Yamaha, however, it turns out that the 719 is just a re-badged
-715, and the chipsets are identical.  The chipset detection code
-has been updated to reflect this.
-
-Anyways, all of these chipsets implement the following devices:
-
-OPL3 FM synthesizer
-Soundblaster Pro
-Microsoft/Windows Sound System
-MPU401 MIDI interface
-
-Note that this driver uses the MSS device, and to my knowledge these
-chipsets enforce an either/or situation with the Soundblaster Pro
-device and the MSS device.  Since the MSS device has better
-capabilities, I have implemented the driver to use it.
-
-
-Mixer Channels
---------------
-
-Older versions of this driver (pre-December 2000) had two mixers,
-an OPL3-SA2 or SA3 mixer and a MSS mixer.  The OPL3-SA[23] mixer
-device contained a superset of mixer channels consisting of its own
-channels and all of the MSS mixer channels.  To simplify the driver
-considerably, and to partition functionality better, the OPL3-SA[23]
-mixer device now contains has its own specific mixer channels.  They
-are:
-
-Volume     - Hardware master volume control
-Bass       - SA3 only, now supports left and right channels
-Treble     - SA3 only, now supports left and right channels
-Microphone - Hardware microphone input volume control
-Digital1   - Yamaha 3D enhancement "Wide" mixer
-
-All other mixer channels (e.g. "PCM", "CD", etc.) now have to be
-controlled via the "MS Sound System (CS4231)" mixer.  To facilitate
-this, the mixer device creation order has been switched so that
-the MSS mixer is created first.  This allows accessing the majority
-of the useful mixer channels even via single mixer-aware tools
-such as "aumix".
-
-
-Plug 'n Play
-------------
-
-In previous kernels (2.2.x), some configuration was required to
-get the driver to talk to the card.  Being the new millennium and
-all, the 2.4.x kernels now support auto-configuration if ISA PnP
-support is configured in.  Theoretically, the driver even supports
-having more than one card in this case.
-
-With the addition of PnP support to the driver, two new parameters
-have been added to control it:
-
-isapnp   - set to 0 to disable ISA PnP card detection
-
-multiple - set to 0 to disable multiple PnP card detection
-
-
-Optional Parameters
--------------------
-
-Recent (December 2000) additions to the driver (based on a patch
-provided by Peter Englmaier) are two new parameters:
-
-ymode -    Set Yamaha 3D enhancement mode:
-           0 = Desktop/Normal  5-12 cm speakers
-           1 = Notebook PC (1) 3 cm speakers
-           2 = Notebook PC (2) 1.5 cm speakers
-           3 = Hi-Fi           16-38 cm speakers
-
-loopback - Set A/D input source. Useful for echo cancellation:
-           0 = Mic Right channel (default)
-           1 = Mono output loopback
-
-The ymode parameter has been tested and does work.  The loopback
-parameter, however, is untested.  Any feedback on its usefulness
-would be appreciated.
-
-
-Manual Configuration
---------------------
-
-If for some reason you decide not to compile ISA PnP support into
-your kernel, or disabled the driver's usage of it by setting the
-isapnp parameter as discussed above, then you will need to do some
-manual configuration.  There are two ways of doing this.  The most
-common is to use the isapnptools package to initialize the card, and
-use the kernel module form of the sound subsystem and sound drivers.
-Alternatively, some BIOS's allow manual configuration of installed
-PnP devices in a BIOS menu, which should allow using the non-modular
-sound drivers, i.e. built into the kernel.
-
-I personally use isapnp and modules, and do not have access to a PnP
-BIOS machine to test.  If you have such a beast, configuring the
-driver to be built into the kernel should just work (thanks to work
-done by David Luyer <luyer@ucs.uwa.edu.au>).  You will still need
-to specify settings, which can be done by adding:
-
-opl3sa2=<io>,<irq>,<dma>,<dma2>,<mssio>,<mpuio>
-
-to the kernel command line.  For example:
-
-opl3sa2=0x370,5,0,1,0x530,0x330
-
-If you are instead using the isapnp tools (as most people have been
-before Linux 2.4.x), follow the directions in their documentation to
-produce a configuration file.  Here is the relevant excerpt I used to
-use for my SA3 card from my isapnp.conf:
-
-(CONFIGURE YMH0800/-1 (LD 0
-
-# NOTE: IO 0 is for the unused SoundBlaster part of the chipset.
-(IO 0 (BASE 0x0220))
-(IO 1 (BASE 0x0530))
-(IO 2 (BASE 0x0388))
-(IO 3 (BASE 0x0330))
-(IO 4 (BASE 0x0370))
-(INT 0 (IRQ 5 (MODE +E)))
-(DMA 0 (CHANNEL 0))
-(DMA 1 (CHANNEL 1))
-
-Here, note that:
-
-Port  Acceptable Range  Purpose
-----  ----------------  -------
-IO 0  0x0220 - 0x0280   SB base address, unused.
-IO 1  0x0530 - 0x0F48   MSS base address
-IO 2  0x0388 - 0x03F8   OPL3 base address
-IO 3  0x0300 - 0x0334   MPU base address
-IO 4  0x0100 - 0x0FFE   card's own base address for its control I/O ports
-
-The IRQ and DMA values can be any that are considered acceptable for a
-MSS.  Assuming you've got isapnp all happy, then you should be able to
-do something like the following (which matches up with the isapnp
-configuration above):
-
-modprobe mpu401
-modprobe ad1848
-modprobe opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=5 dma=0 dma2=1
-modprobe opl3 io=0x388
-
-See the section "Automatic Module Loading" below for how to set up
-/etc/modprobe.conf to automate this.
-
-An important thing to remember that the opl3sa2 module's io argument is
-for it's own control port, which handles the card's master mixer for
-volume (on all cards), and bass and treble (on SA3 cards).
-
-
-Troubleshooting
----------------
-
-If all goes well and you see no error messages, you should be able to
-start using the sound capabilities of your system.  If you get an
-error message while trying to insert the opl3sa2 module, then make
-sure that the values of the various arguments match what you specified
-in your isapnp configuration file, and that there is no conflict with
-another device for an I/O port or interrupt.  Checking the contents of
-/proc/ioports and /proc/interrupts can be useful to see if you're
-butting heads with another device.
-
-If you still cannot get the module to load, look at the contents of
-your system log file, usually /var/log/messages.  If you see the
-message "opl3sa2: Unknown Yamaha audio controller version", then you
-have a different chipset version than I've encountered so far.  Look
-for all messages in the log file that start with "opl3sa2: " and see
-if they provide any clues.  If you do not see the chipset version
-message, and none of the other messages present in the system log are
-helpful, email me some details and I'll try my best to help.
-
-
-Automatic Module Loading
-------------------------
-
-Lastly, if you're using modules and want to set up automatic module
-loading with kmod, the kernel module loader, here is the section I
-currently use in my modprobe.conf file:
-
-# Sound
-alias sound-slot-0 opl3sa2
-options opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=7 dma=0 dma2=3
-options opl3 io=0x388
-
-That's all it currently takes to get an OPL3-SA3 card working on my
-system.  Once again, if you have any other problems, email me at the
-address listed above.
-
-Scott
diff --git a/Documentation/sound/oss/VIA-chipset b/Documentation/sound/oss/VIA-chipset
deleted file mode 100644
index 37865234e54d..000000000000
--- a/Documentation/sound/oss/VIA-chipset
+++ /dev/null
@@ -1,43 +0,0 @@
-Running sound cards on VIA chipsets
-
-o	There are problems with VIA chipsets and sound cards that appear to
-	lock the hardware solidly. Test programs under DOS have verified the
-	problem exists on at least some (but apparently not all) VIA boards
-
-o	VIA have so far failed to bother to answer support mail on the subject
-	so if you are a VIA engineer feeling aggrieved as you read this 
-	document go chase your own people. If there is a workaround please
-	let us know so we can implement it.
-
-
-Certain patterns of ISA DMA access used for most PC sound cards cause the
-VIA chipsets to lock up. From the collected reports this appears to cover a
-wide range of boards. Some also lock up with sound cards under Win* as well.
-
-Linux implements a workaround providing your chipset is PCI and you compiled
-with PCI Quirks enabled. If so you will see a message
-	"Activating ISA DMA bug workarounds"
-
-during booting. If you have a VIA PCI chipset that hangs when you use the
-sound and is not generating this message even with PCI quirks enabled
-please report the information to the linux-kernel list (see REPORTING-BUGS).
-
-If you are one of the tiny number of unfortunates with a 486 ISA/VLB VIA
-chipset board you need to do the following to build a special kernel for
-your board
-
-	edit linux/include/asm-i386/dma.h
-
-change
-
-#define isa_dma_bridge_buggy 	(0)
-
-to
-
-#define isa_dma_bridge_buggy 	(1)
-
-and rebuild a kernel without PCI quirk support.
-
-
-Other than this particular glitch the VIA [M]VP* chipsets appear to work
-perfectly with Linux.
diff --git a/Documentation/sound/oss/cs46xx b/Documentation/sound/oss/cs46xx
deleted file mode 100644
index b54432709863..000000000000
--- a/Documentation/sound/oss/cs46xx
+++ /dev/null
@@ -1,138 +0,0 @@
-
-Documentation for the Cirrus Logic/Crystal SoundFusion cs46xx/cs4280 audio 
-controller chips (2001/05/11)
-
-The cs46xx audio driver supports the DSP line of Cirrus controllers. 
-Specifically, the cs4610, cs4612, cs4614, cs4622, cs4624, cs4630 and the cs4280
-products.  This driver uses the generic ac97_codec driver for AC97 codec
-support.  
-
-
-Features:
-
-Full Duplex Playback/Capture supported from 8k-48k.
-16Bit Signed LE & 8Bit Unsigned, with Mono or Stereo supported.
-
-APM/PM - 2.2.x PM is enabled and functional. APM can also
-be enabled for 2.4.x by modifying the CS46XX_ACPI_SUPPORT macro
-definition.
-
-DMA playback buffer size is configurable from 16k (defaultorder=2) up to 2Meg 
-(defaultorder=11).  DMA capture buffer size is fixed at a single 4k page as
-two 2k fragments.
-
-MMAP seems to work well with QuakeIII, and test XMMS plugin.
-
-Myth2 works, but the polling logic is not fully correct, but is functional.
-
-The 2.4.4-ac6 gameport code in the cs461x joystick driver has been tested 
-with a Microsoft Sidewinder joystick (cs461x.o and sidewinder.o).  This 
-audio driver must be loaded prior to the joystick driver to enable the
-DSP task image supporting the joystick device.
-
-
-Limitations:
-
-SPDIF is currently not supported.
-
-Primary codec support only.  No secondary codec support is implemented.
-
-
-
-NOTES:
-
-Hercules Game Theatre XP - the EGPIO2 pin controls the external Amp,
-and has been tested.
-Module parameter hercules_egpio_disable set to 1, will force a 0 to EGPIODR
-to disable the external amplifier.
-
-VTB Santa Cruz - the GPIO7/GPIO8 on the Secondary Codec control
-the external amplifier for the "back" speakers, since we do not
-support the secondary codec then this external amp is not
-turned on.  The primary codec external amplifier is supported but
-note that the AC97 EAPD bit is inverted logic (amp_voyetra()).
-
-DMA buffer size - there are issues with many of the Linux applications
-concerning the optimal buffer size.  Several applications request a 
-certain fragment size and number and then do not verify that the driver
-has the ability to support the requested configuration.  
-SNDCTL_DSP_SETFRAGMENT ioctl is used to request a fragment size and
-number of fragments.  Some applications exit if an error is returned
-on this particular ioctl. Therefore, in alignment with the other OSS audio 
-drivers, no error is returned when a SETFRAGs IOCTL is received, but the 
-values passed from the app are not used in any buffer calculation 
-(ossfragshift/ossmaxfrags are not used).
-Use the "defaultorder=N" module parameter to change the buffer size if
-you have an application that requires a specific number of fragments
-or a specific buffer size (see below).
-
-Debug Interface
----------------
-There is an ioctl debug interface to allow runtime modification of the 
-debug print levels.  This debug interface code can be disabled from the 
-compilation process with commenting the following define:
-#define CSDEBUG_INTERFACE 1
-There is also a debug print methodolgy to select printf statements from
-different areas of the driver.  A debug print level is also used to allow
-additional printfs to be active.  Comment out the following line in the
-driver to disable compilation of the CS_DBGOUT print statements:
-#define CSDEBUG 1
- 
-Please see the definitions for cs_debuglevel and cs_debugmask for additional
-information on the debug levels and sections.
-
-There is also a csdbg executable to allow runtime manipulation of these 
-parameters.  for a copy email: twoller@crystal.cirrus.com
-
-
-
-MODULE_PARMS definitions
-------------------------
-module_param(defaultorder, ulong, 0);
-defaultorder=N
-where N is a value from 1 to 12
-The buffer order determines the size of the dma buffer for the driver.
-under Linux, a smaller buffer allows more responsiveness from many of the 
-applications (e.g. games).  A larger buffer allows some of the apps (esound) 
-to not underrun the dma buffer as easily.  As default, use 32k (order=3)
-rather than 64k as some of the games work more responsively.
-(2^N) * PAGE_SIZE = allocated buffer size
-
-module_param(cs_debuglevel, ulong, 0644);
-module_param(cs_debugmask, ulong, 0644);
-cs_debuglevel=N
-cs_debugmask=0xMMMMMMMM
-where N is a value from 0 (no debug printfs), to 9 (maximum)
-0xMMMMMMMM is a debug mask corresponding to the CS_xxx bits (see driver source).
-
-module_param(hercules_egpio_disable, ulong, 0);
-hercules_egpio_disable=N
-where N is a 0 (enable egpio), or a 1 (disable egpio support)
-
-module_param(initdelay, ulong, 0);
-initdelay=N
-This value is used to determine the millescond delay during the initialization
-code prior to powering up the PLL.  On laptops this value can be used to
-assist with errors on resume, mostly with IBM laptops.  Basically, if the 
-system is booted under battery power then the mdelay()/udelay() functions fail to 
-properly delay the required time.  Also, if the system is booted under AC power
-and then the power removed, the mdelay()/udelay() functions will not delay properly.
- 
-module_param(powerdown, ulong, 0);
-powerdown=N
-where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
-
-
-module_param(external_amp, bool, 0);
-external_amp=1
-if N is set to 1, then force enabling the EAPD support in the primary AC97 codec.
-override the detection logic and force the external amp bit in the AC97 0x26 register
-to be reset (0).  EAPD should be 0 for powerup, and 1 for powerdown.  The VTB Santa Cruz
-card has inverted logic, so there is a special function for these cards.
-
-module_param(thinkpad, bool, 0);
-thinkpad=1
-if N is set to 1, then force enabling the clkrun functionality.
-Currently, when the part is being used, then clkrun is disabled for the entire system,
-but re-enabled when the driver is released or there is no outstanding open count.
-