Merge git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa

* git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (124 commits)
  [ALSA] version 1.0.11rc4
  [PATCH] Intruduce DMA_28BIT_MASK
  [ALSA] hda-codec - Add support for ASUS P4GPL-X
  [ALSA] hda-codec - Add support for HP nx9420 laptop
  [ALSA] Fix memory leaks in error path of control.c
  [ALSA] AMD Au1x00: AC'97 controller is memory mapped
  [ALSA] AMD Au1x00: fix DMA init/cleanup
  [ALSA] hda-codec - Fix generic auto-configurator
  [ALSA] hda-codec - Fix BIOS auto-configuration
  [ALSA] Fixes typos in Audiophile-USB.txt
  [ALSA] ice1712 - typo fixes for dxr_enable module option
  [ALSA] AMD Au1x00: make driver build after cleanup
  [ALSA] ice1712 - Fix wrong value types for enum items
  [ALSA] fix resource leak in usbmixer
  [ALSA] Fix gus_pcm dereference before NULL
  [ALSA] Fix seq_clientmgr dereferences before NULL check
  [ALSA] hda-codec - Fix for Samsung R65 and ASUS A6J
  [ALSA] hda-codec - Add support for VAIO FE550G and SZ110
  [ALSA] usb-audio: add Maya44 mixer control names
  [ALSA] usb-audio: add Casio PL-40R support
  ...

3 files changed, 404 insertions, 6 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 36b511c7cade..1def6049784c 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -513,6 +513,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-ens1371
   ------------------
 
@@ -526,6 +528,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards and autoprobe.
     
+    The power-management is supported.
+
   Module snd-es968
   ----------------
 
@@ -671,6 +675,8 @@ 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)
+    single_cmd  - Use single immediate commands to communicate with
+		codecs (for debugging only)
 
     This module supports one card and autoprobe.
 
@@ -694,13 +700,34 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  asus		3-jack
 	  uniwill	3-jack
 	  F1734		2-jack
+	  lg		LG laptop (m1 express dual)
 	  test		for testing/debugging purpose, almost all controls can be
 			adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y
+	  auto		auto-config reading BIOS (default)
 
 	ALC260
 	  hp		HP machines
 	  fujitsu	Fujitsu S7020
+	  acer		Acer TravelMate
+	  basic		fixed pin assignment (old default model)
+	  auto		auto-config reading BIOS (default)
+
+	ALC262
+	  fujitsu	Fujitsu Laptop
+	  basic		fixed pin assignment w/o SPDIF
+	  auto		auto-config reading BIOS (default)
+
+	ALC882/883/885
+	  3stack-dig	3-jack with SPDIF I/O
+	  6stck-dig	6-jack digital with SPDIF I/O
+	  auto		auto-config reading BIOS (default)
+
+	ALC861
+	  3stack	3-jack
+	  3stack-dig	3-jack with SPDIF I/O
+	  6stack-dig	6-jack with SPDIF I/O
+	  auto		auto-config reading BIOS (default)
 
 	CMI9880
 	  minimal	3-jack in back
@@ -710,6 +737,28 @@ 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)
 
+	AD1981
+	  basic		3-jack (default)
+	  hp		HP nx6320
+
+	AD1986A
+	  6stack	6-jack, separate surrounds (default)
+	  3stack	3-stack, shared surrounds
+	  laptop	2-channel only (FSC V2060, Samsung M50)
+	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)
+
+	AD1988
+	  6stack	6-jack
+	  6stack-dig	ditto with SPDIF
+	  3stack	3-jack
+	  3stack-dig	ditto with SPDIF
+	  laptop	3-jack with hp-jack automute
+	  laptop-dig	ditto with SPDIF
+	  auto		auto-confgi reading BIOS (default)
+
+	STAC7661(?)
+	  vaio		Setup for VAIO FE550G/SZ110
+
     If the default configuration doesn't work and one of the above
     matches with your device, report it together with the PCI
     subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel
@@ -723,6 +772,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	    (Usually SD_LPLIB register is more accurate than the
 	    position buffer.)
 
+    NB: If you get many "azx_get_response timeout" messages at
+    loading, it's likely a problem of interrupts (e.g. ACPI irq
+    routing).  Try to boot with options like "pci=noacpi".  Also, you
+    can try "single_cmd=1" module option.  This will switch the
+    communication method between HDA controller and codecs to the
+    single immediate commands instead of CORB/RIRB.  Basically, the
+    single command mode is provided only for BIOS, and you won't get
+    unsolicited events, too.  But, at least, this works independently
+    from the irq.  Remember this is a last resort, and should be
+    avoided as much as possible...
+    
     The power-management is supported.
 
   Module snd-hdsp
@@ -802,6 +862,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   ------------------
 
     Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+			* MidiMan M Audio Revolution 5.1
 			* MidiMan M Audio Revolution 7.1
 			* AMP Ltd AUDIO2000
 			* TerraTec Aureon 5.1 Sky
@@ -810,6 +871,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			* TerraTec Phase 22
 			* TerraTec Phase 28
 			* AudioTrak Prodigy 7.1
+			* AudioTrak Prodigy 7.1LT
 			* AudioTrak Prodigy 192
 			* Pontis MS300
 			* Albatron K8X800 Pro II 
@@ -820,9 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			* Shuttle SN25P
 
     model       - Use the given board model, one of the following:
-		  revo71, amp2000, prodigy71, prodigy192, aureon51,
-		  aureon71, universe, k8x800, phase22, phase28, ms300,
-		  av710
+		  revo51, revo71, amp2000, prodigy71, prodigy71lt,
+		  prodigy192, aureon51, aureon71, universe,
+		  k8x800, phase22, phase28, ms300, av710
 
     This module supports multiple cards and autoprobe.
 
@@ -1353,6 +1415,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     vid             - Vendor ID for the device (optional)
     pid             - Product ID for the device (optional)
+    device_setup    - Device specific magic number (optional)
+                    - Influence depends on the device
+                    - Default: 0x0000 
 
     This module supports multiple devices, autoprobe and hotplugging.
 
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
new file mode 100644
index 000000000000..4692c8e77dc1
--- /dev/null
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -0,0 +1,333 @@
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.2
+	========================================================
+
+	    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.
+
+1 - Audiophile USB Specs and correct usage
+==========================================
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+ * Analog Stereo Input (Ai)
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following caracteristics:
+* 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"
+
+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 additionnaly supports surround-encoded AC-3 and DTS passthrough, 
+though I haven't tested it under linux
+   - Note that in this setup only the Do interface can be enabled
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+to synchronize the device to an external sample clock
+   - As a consequence, the Di port must be enable only if an active Digital 
+source is connected
+   - Enabling Di when no digital source is connected can result in a 
+synchronization error (for instance sound played at an odd sample rate)
+
+
+2 - Audiophile USB support in ALSA
+==================================
+
+2.1 - MIDI ports
+----------------
+The Audiophile USB MIDI ports will be automatically supported once the 
+following modules have been loaded:
+ * snd-usb-audio
+ * snd-seq
+ * snd-seq-midi
+
+No additionnal setting is required.
+
+2.2 - Audio ports
+-----------------
+
+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 behaviour 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 sample rates and any sample depths supported). 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):
+ * 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.
+
+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
+
+If you're happy with the default Alsa driver setup and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+2.2.2 - 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).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called "device_setup".
+
+2.2.2.1 - Initializing the working mode of the Audiohile USB
+
+As far as the Audiohile USB device is concerned, this value let the user 
+specify:
+ * the sample depth
+ * 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 decribed earlier
+ * 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
+ * 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
+ * device_setup=0x03
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru (not tested)
+   - 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 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_ intialize 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:
+   - first trun off the device
+   - de-register the snd-usb-audio module
+   - change the device_setup parameter (by either manually reprobing the module 
+     or changing modprobe.conf)
+   - turn on the device
+
+2.2.2.3 - 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 thi 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
+
+2.2.3 -  USB implementation details for this device
+
+You may safely skip this section if you're not interrested in driver 
+development.
+
+This section describes some internals aspect of the device and summarize 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 intialization of the device, the driver 
+_must_know_ how the device will be used:
+ * if DTS is choosen, 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_enpoint 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
+===================================
+
+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
+--------------------------------
+
+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.
+
+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
+
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index 4251085d38d3..6dc9d9f622ca 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1834,7 +1834,7 @@
           mychip_set_sample_format(chip, runtime->format);
           mychip_set_sample_rate(chip, runtime->rate);
           mychip_set_channels(chip, runtime->channels);
-          mychip_set_dma_setup(chip, runtime->dma_area,
+          mychip_set_dma_setup(chip, runtime->dma_addr,
                                chip->buffer_size,
                                chip->period_size);
           return 0;
@@ -3388,7 +3388,7 @@ struct _snd_pcm_runtime {
           .name = "PCM Playback Switch",
           .index = 0,
           .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
-          .private_values = 0xffff,
+          .private_value = 0xffff,
           .info = my_control_info,
           .get = my_control_get,
           .put = my_control_put
@@ -3449,7 +3449,7 @@ struct _snd_pcm_runtime {
       </para>
 
       <para>
-        The <structfield>private_values</structfield> field contains
+        The <structfield>private_value</structfield> field contains
       an arbitrary long integer value for this record. When using
       generic <structfield>info</structfield>,
       <structfield>get</structfield> and