r2398 - branches/src/target/kernel/2.6.22.x/patches

shoragan at sita.openmoko.org shoragan at sita.openmoko.org
Thu Jul 26 17:24:39 CEST 2007


Author: shoragan
Date: 2007-07-26 17:24:28 +0200 (Thu, 26 Jul 2007)
New Revision: 2398

Added:
   branches/src/target/kernel/2.6.22.x/patches/alsa-2.6.23-rc1-commit.diff
   branches/src/target/kernel/2.6.22.x/patches/asoc-kconfig-fix.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c2410_udc_from_upstream.patch
Removed:
   branches/src/target/kernel/2.6.22.x/patches/asoc-asm_hardware_h.patch
   branches/src/target/kernel/2.6.22.x/patches/asoc.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c2410-usb-switch.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c2410_udc-vbus_draw_pdata.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c2410_udc.patch
   branches/src/target/kernel/2.6.22.x/patches/series.old
Modified:
   branches/src/target/kernel/2.6.22.x/patches/gta01-core.patch
   branches/src/target/kernel/2.6.22.x/patches/gta01-no_nand_partitions.patch
   branches/src/target/kernel/2.6.22.x/patches/gta01-pcf50606.patch
   branches/src/target/kernel/2.6.22.x/patches/gta02-core.patch
   branches/src/target/kernel/2.6.22.x/patches/hxd8-core.patch
   branches/src/target/kernel/2.6.22.x/patches/qt2410-s3c_mci-pdata.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c_mci.patch
   branches/src/target/kernel/2.6.22.x/patches/s3c_mci_platform.patch
   branches/src/target/kernel/2.6.22.x/patches/s3cmci-dma-free.patch
   branches/src/target/kernel/2.6.22.x/patches/s3cmci-stop-fix.patch
   branches/src/target/kernel/2.6.22.x/patches/s3cmci_dbg.patch
   branches/src/target/kernel/2.6.22.x/patches/series
   branches/src/target/kernel/2.6.22.x/patches/smedia-glamo.patch
Log:
Port patches to 2.6.22.1


Added: branches/src/target/kernel/2.6.22.x/patches/alsa-2.6.23-rc1-commit.diff
===================================================================
--- branches/src/target/kernel/2.6.22.x/patches/alsa-2.6.23-rc1-commit.diff	2007-07-26 14:40:11 UTC (rev 2397)
+++ branches/src/target/kernel/2.6.22.x/patches/alsa-2.6.23-rc1-commit.diff	2007-07-26 15:24:28 UTC (rev 2398)
@@ -0,0 +1,10960 @@
+--- linux-2.6.22.1.orig/CREDITS
++++ linux-2.6.22.1/CREDITS
+@@ -2212,13 +2212,13 @@
+ S: Denmark
+ 
+ N: Claudio S. Matsuoka
+-E: claudio at conectiva.com
+-E: claudio at helllabs.org
++E: cmatsuoka at gmail.com
++E: claudio at mandriva.com
+ W: http://helllabs.org/~claudio
+-D: V4L, OV511 driver hacks
++D: V4L, OV511 and HDA-codec hacks
+ S: Conectiva S.A.
+-S: R. Tocantins 89
+-S: 80050-430  Curitiba PR
++S: Souza Naves 1250
++S: 80050-040  Curitiba PR
+ S: Brazil
+ 
+ N: Heinz Mauelshagen
+--- linux-2.6.22.1.orig/Documentation/sound/alsa/ALSA-Configuration.txt
++++ linux-2.6.22.1/Documentation/sound/alsa/ALSA-Configuration.txt
+@@ -467,7 +467,12 @@
+     above explicitly.
+ 
+     The power-management is supported.
+-    
++
++  Module snd-cs5530
++  _________________
++
++    Module for Cyrix/NatSemi Geode 5530 chip. 
++  
+   Module snd-cs5535audio
+   ----------------------
+ 
+@@ -759,6 +764,7 @@
+ 
+     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 @@
+ 	  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 @@
+ 	  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 @@
+ 	  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 @@
+ 	  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 @@
+ 	  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 @@
+ 	  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 @@
+     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
+--- linux-2.6.22.1.orig/Documentation/sound/alsa/Audiophile-Usb.txt
++++ linux-2.6.22.1/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 at 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 at 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 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 @@
+ 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.
++3.1 - Default Alsa driver mode
++------------------------------
+ 
+-In this case the Audiophile ports are mapped to alsa pcm devices in the 
+-following way (I suppose the device's index is 1):
++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 @@
+    % 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 @@
+  * 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 @@
+    - 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 @@
+    - 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 @@
+ "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 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 ;-) ).
+ 
+-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.
++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 @@
+ * 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 @@
+ 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.
+--- linux-2.6.22.1.orig/Documentation/sound/alsa/OSS-Emulation.txt
++++ linux-2.6.22.1/Documentation/sound/alsa/OSS-Emulation.txt
+@@ -278,6 +278,21 @@
+ 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
+ ====================
+ 
+--- linux-2.6.22.1.orig/include/linux/i2c-id.h
++++ linux-2.6.22.1/include/linux/i2c-id.h
+@@ -115,9 +115,10 @@
+ #define I2C_DRIVERID_KS0127	86	/* Samsung ks0127 video decoder */
+ #define I2C_DRIVERID_TLV320AIC23B 87	/* TI TLV320AIC23B audio codec  */
+ #define I2C_DRIVERID_ISL1208	88	/* Intersil ISL1208 RTC		*/
+-#define I2C_DRIVERID_WM8731		89	/* Wolfson WM8731 audio codec */
+-#define I2C_DRIVERID_WM8750		90	/* Wolfson WM8750 audio codec */
+-#define I2C_DRIVERID_WM8753		91	/* Wolfson WM8753 audio codec */
++#define I2C_DRIVERID_WM8731	89	/* Wolfson WM8731 audio codec */
++#define I2C_DRIVERID_WM8750	90	/* Wolfson WM8750 audio codec */
++#define I2C_DRIVERID_WM8753	91	/* Wolfson WM8753 audio codec */
++#define I2C_DRIVERID_LM4857 	92 	/* LM4857 Audio Amplifier */
+ 
+ #define I2C_DRIVERID_I2CDEV	900
+ #define I2C_DRIVERID_ARP        902    /* SMBus ARP Client              */
+--- linux-2.6.22.1.orig/include/sound/ak4xxx-adda.h
++++ linux-2.6.22.1/include/sound/ak4xxx-adda.h
+@@ -43,6 +43,7 @@
+ struct snd_akm4xxx_dac_channel {
+ 	char *name;		/* mixer volume name */
+ 	unsigned int num_channels;
++	char *switch_name;		/* mixer switch*/
+ };
+ 
+ /* ADC labels and channels */
+--- linux-2.6.22.1.orig/include/sound/cs46xx.h
++++ linux-2.6.22.1/include/sound/cs46xx.h
+@@ -1723,6 +1723,10 @@
+ 	struct snd_cs46xx_pcm *playback_pcm;
+ 	unsigned int play_ctl;
+ #endif
++
++#ifdef CONFIG_PM
++	u32 *saved_regs;
++#endif
+ };
+ 
+ int snd_cs46xx_create(struct snd_card *card,
+--- linux-2.6.22.1.orig/include/sound/cs46xx_dsp_spos.h
++++ linux-2.6.22.1/include/sound/cs46xx_dsp_spos.h
+@@ -107,6 +107,7 @@
+ 	char scb_name[DSP_MAX_SCB_NAME];
+ 	u32 address;
+ 	int index;
++	u32 *data;
+ 
+ 	struct dsp_scb_descriptor * sub_list_ptr;
+ 	struct dsp_scb_descriptor * next_scb_ptr;
+@@ -127,6 +128,7 @@
+ 	int size;
+ 	u32 address;
+ 	int index;
++	u32 *data;
+ };
+ 
+ struct dsp_pcm_channel_descriptor {
+--- linux-2.6.22.1.orig/include/sound/emu10k1.h
++++ linux-2.6.22.1/include/sound/emu10k1.h
+@@ -1120,6 +1120,16 @@
+ /************************************************************************************************/
+ /* EMU1010m HANA Destinations									*/
+ /************************************************************************************************/
++/* 32-bit destinations of signal in the Hana FPGA. Destinations are either
++ * physical outputs of Hana, or outputs going to Alice2 (audigy) for capture
++ * - 16 x EMU_DST_ALICE2_EMU32_X.
++ */
++/* EMU32 = 32-bit serial channel between Alice2 (audigy) and Hana (FPGA) */
++/* EMU_DST_ALICE2_EMU32_X - data channels from Hana to Alice2 used for capture.
++ * Which data is fed into a EMU_DST_ALICE2_EMU32_X channel in Hana depends on
++ * setup of mixer control for each destination - see emumixer.c -
++ * snd_emu1010_output_enum_ctls[], snd_emu1010_input_enum_ctls[]
++ */
+ #define EMU_DST_ALICE2_EMU32_0	0x000f	/* 16 EMU32 channels to Alice2 +0 to +0xf */
+ #define EMU_DST_ALICE2_EMU32_1	0x0000	/* 16 EMU32 channels to Alice2 +0 to +0xf */
+ #define EMU_DST_ALICE2_EMU32_2	0x0001	/* 16 EMU32 channels to Alice2 +0 to +0xf */
+@@ -1199,6 +1209,12 @@
+ /************************************************************************************************/
+ /* EMU1010m HANA Sources									*/
+ /************************************************************************************************/
++/* 32-bit sources of signal in the Hana FPGA. The sources are routed to
++ * destinations using mixer control for each destination - see emumixer.c
++ * Sources are either physical inputs of FPGA,
++ * or outputs from Alice (audigy) - 16 x EMU_SRC_ALICE_EMU32A +
++ * 16 x EMU_SRC_ALICE_EMU32B
++ */
+ #define EMU_SRC_SILENCE		0x0000	/* Silence */
+ #define EMU_SRC_DOCK_MIC_A1	0x0100	/* Audio Dock Mic A, 1st or 48kHz only */
+ #define EMU_SRC_DOCK_MIC_A2	0x0101	/* Audio Dock Mic A, 2nd or 96kHz */
+--- linux-2.6.22.1.orig/include/sound/sb.h
++++ linux-2.6.22.1/include/sound/sb.h
+@@ -38,6 +38,7 @@
+ 	SB_HW_ALS100,		/* Avance Logic ALS100 chip */
+ 	SB_HW_ALS4000,		/* Avance Logic ALS4000 chip */
+ 	SB_HW_DT019X,		/* Diamond Tech. DT-019X / Avance Logic ALS-007 */
++	SB_HW_CS5530,		/* Cyrix/NatSemi 5530 VSA1 */
+ };
+ 
+ #define SB_OPEN_PCM			0x01
+--- linux-2.6.22.1.orig/include/sound/version.h
++++ linux-2.6.22.1/include/sound/version.h
+@@ -1,3 +1,3 @@
+ /* include/version.h.  Generated by alsa/ksync script.  */
+ #define CONFIG_SND_VERSION "1.0.14"
+-#define CONFIG_SND_DATE " (Thu May 31 09:03:25 2007 UTC)"
++#define CONFIG_SND_DATE " (Fri Jul 20 09:12:58 2007 UTC)"
+--- linux-2.6.22.1.orig/include/sound/wavefront_fx.h
++++ /dev/null
+@@ -1,9 +0,0 @@
+-#ifndef __SOUND_WAVEFRONT_FX_H
+-#define __SOUND_WAVEFRONT_FX_H
+-
+-extern int  snd_wavefront_fx_detect (snd_wavefront_t *);
+-extern void snd_wavefront_fx_ioctl  (snd_synth_t *sdev, 
+-				     unsigned int cmd, 
+-				     unsigned long arg);
+-
+-#endif  __SOUND_WAVEFRONT_FX_H
+--- linux-2.6.22.1.orig/sound/Kconfig
++++ linux-2.6.22.1/sound/Kconfig
+@@ -65,6 +65,8 @@
+ 
+ source "sound/mips/Kconfig"
+ 
++source "sound/sh/Kconfig"
++
+ # the following will depend on the order of config.
+ # here assuming USB is defined before ALSA
+ source "sound/usb/Kconfig"
+--- linux-2.6.22.1.orig/sound/Makefile
++++ linux-2.6.22.1/sound/Makefile
+@@ -5,7 +5,7 @@
+ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
+ obj-$(CONFIG_SOUND_PRIME) += oss/
+ obj-$(CONFIG_DMASOUND) += oss/
+-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
++obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
+ obj-$(CONFIG_SND_AOA) += aoa/
+ 
+ # This one must be compilable even if sound is configured out
+--- linux-2.6.22.1.orig/sound/aoa/codecs/snd-aoa-codec-onyx.c
++++ linux-2.6.22.1/sound/aoa/codecs/snd-aoa-codec-onyx.c
+@@ -661,7 +661,7 @@
+ 		.tag = 2,
+ 	},
+ #ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+-Once alsa gets supports for this kind of thing we can add it...
++	/* Once alsa gets supports for this kind of thing we can add it... */
+ 	{
+ 		/* digital compressed output */
+ 		.formats =  SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
+@@ -713,7 +713,7 @@
+ 	if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
+ 		/* mute and lock analog output */
+ 		onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+-		if (onyx_write_register(onyx
++		if (onyx_write_register(onyx,
+ 					ONYX_REG_DAC_CONTROL,
+ 					v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
+ 			goto out_unlock;
+--- linux-2.6.22.1.orig/sound/core/pcm_native.c
++++ linux-2.6.22.1/sound/core/pcm_native.c
+@@ -1487,7 +1487,7 @@
+ 
+ 	snd_pcm_stream_lock_irq(substream);
+ 	/* resume pause */
+-	if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
++	if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+ 		snd_pcm_pause(substream, 0);
+ 
+ 	/* pre-start/stop - all running streams are changed to DRAINING state */
+--- linux-2.6.22.1.orig/sound/core/seq/seq_instr.c
++++ linux-2.6.22.1/sound/core/seq/seq_instr.c
+@@ -109,7 +109,7 @@
+ 			spin_lock_irqsave(&list->lock, flags);
+ 			while (instr->use) {
+ 				spin_unlock_irqrestore(&list->lock, flags);
+-				schedule_timeout_interruptible(1);
++				schedule_timeout(1);
+ 				spin_lock_irqsave(&list->lock, flags);
+ 			}				
+ 			spin_unlock_irqrestore(&list->lock, flags);
+@@ -199,7 +199,7 @@
+ 			instr = flist;
+ 			flist = instr->next;
+ 			while (instr->use)
+-				schedule_timeout_interruptible(1);
++				schedule_timeout(1);
+ 			if (snd_seq_instr_free(instr, atomic)<0)
+ 				snd_printk(KERN_WARNING "instrument free problem\n");
+ 			instr = next;
+@@ -555,7 +555,7 @@
+ 					   SNDRV_SEQ_INSTR_NOTIFY_REMOVE);
+ 		while (instr->use) {
+ 			spin_unlock_irqrestore(&list->lock, flags);
+-			schedule_timeout_interruptible(1);
++			schedule_timeout(1);
+ 			spin_lock_irqsave(&list->lock, flags);
+ 		}				
+ 		spin_unlock_irqrestore(&list->lock, flags);
+--- linux-2.6.22.1.orig/sound/core/timer.c
++++ linux-2.6.22.1/sound/core/timer.c
+@@ -1549,9 +1549,11 @@
+ 	int err = 0;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	t = tu->timeri->timer;
+-	snd_assert(t != NULL, return -ENXIO);
++	if (!t)
++		return -EBADFD;
+ 
+ 	info = kzalloc(sizeof(*info), GFP_KERNEL);
+ 	if (! info)
+@@ -1579,9 +1581,11 @@
+ 	int err;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	t = tu->timeri->timer;
+-	snd_assert(t != NULL, return -ENXIO);
++	if (!t)
++		return -EBADFD;
+ 	if (copy_from_user(&params, _params, sizeof(params)))
+ 		return -EFAULT;
+ 	if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) {
+@@ -1675,7 +1679,8 @@
+ 	struct snd_timer_status status;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	memset(&status, 0, sizeof(status));
+ 	status.tstamp = tu->tstamp;
+ 	status.resolution = snd_timer_resolution(tu->timeri);
+@@ -1695,7 +1700,8 @@
+ 	struct snd_timer_user *tu;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	snd_timer_stop(tu->timeri);
+ 	tu->timeri->lost = 0;
+ 	tu->last_resolution = 0;
+@@ -1708,7 +1714,8 @@
+ 	struct snd_timer_user *tu;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0;
+ }
+ 
+@@ -1718,7 +1725,8 @@
+ 	struct snd_timer_user *tu;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	tu->timeri->lost = 0;
+ 	return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0;
+ }
+@@ -1729,7 +1737,8 @@
+ 	struct snd_timer_user *tu;
+ 
+ 	tu = file->private_data;
+-	snd_assert(tu->timeri != NULL, return -ENXIO);
++	if (!tu->timeri)
++		return -EBADFD;
+ 	return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0;
+ }
+ 
+--- linux-2.6.22.1.orig/sound/drivers/dummy.c
++++ linux-2.6.22.1/sound/drivers/dummy.c
+@@ -659,7 +659,7 @@
+ 	},
+ };
+ 
+-static void __init_or_module snd_dummy_unregister_all(void)
++static void snd_dummy_unregister_all(void)
+ {
+ 	int i;
+ 
+--- linux-2.6.22.1.orig/sound/drivers/mpu401/mpu401.c
++++ linux-2.6.22.1/sound/drivers/mpu401/mpu401.c
+@@ -228,7 +228,7 @@
+ static struct pnp_driver snd_mpu401_pnp_driver;
+ #endif
+ 
+-static void __init_or_module snd_mpu401_unregister_all(void)
++static void snd_mpu401_unregister_all(void)
+ {
+ 	int i;
+ 
+--- linux-2.6.22.1.orig/sound/drivers/portman2x4.c
++++ linux-2.6.22.1/sound/drivers/portman2x4.c
+@@ -833,7 +833,7 @@
+ /*********************************************************************
+  * module init stuff
+  *********************************************************************/
+-static void __init_or_module snd_portman_unregister_all(void)
++static void snd_portman_unregister_all(void)
+ {
+ 	int i;
+ 
+--- linux-2.6.22.1.orig/sound/drivers/serial-u16550.c
++++ linux-2.6.22.1/sound/drivers/serial-u16550.c
+@@ -998,7 +998,7 @@
+ 	},
+ };
+ 
+-static void __init_or_module snd_serial_unregister_all(void)
++static void snd_serial_unregister_all(void)
+ {
+ 	int i;
+ 
+--- linux-2.6.22.1.orig/sound/drivers/virmidi.c
++++ linux-2.6.22.1/sound/drivers/virmidi.c
+@@ -145,7 +145,7 @@
+ 	},
+ };
+ 
+-static void __init_or_module snd_virmidi_unregister_all(void)
++static void snd_virmidi_unregister_all(void)
+ {
+ 	int i;
+ 
+--- linux-2.6.22.1.orig/sound/i2c/other/ak4xxx-adda.c
++++ linux-2.6.22.1/sound/i2c/other/ak4xxx-adda.c
+@@ -481,8 +481,8 @@
+ 	int addr = AK_GET_ADDR(kcontrol->private_value);
+ 	int shift = AK_GET_SHIFT(kcontrol->private_value);
+ 	int invert = AK_GET_INVERT(kcontrol->private_value);
+-	unsigned char val = snd_akm4xxx_get(ak, chip, addr);
+-
++	/* we observe the (1<<shift) bit only */
++	unsigned char val = snd_akm4xxx_get(ak, chip, addr) & (1<<shift);
+ 	if (invert)
+ 		val = ! val;
+ 	ucontrol->value.integer.value[0] = (val & (1<<shift)) != 0;
+@@ -585,6 +585,26 @@
+ 
+ 	mixer_ch = 0;
+ 	for (idx = 0; idx < ak->num_dacs; ) {
++		/* mute control for Revolution 7.1 - AK4381 */
++		if (ak->type == SND_AK4381 
++				&&  ak->dac_info[mixer_ch].switch_name) {
++			memset(&knew, 0, sizeof(knew));
++			knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
++			knew.count = 1;
++			knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
++			knew.name = ak->dac_info[mixer_ch].switch_name;
++			knew.info = ak4xxx_switch_info;
++			knew.get = ak4xxx_switch_get;
++			knew.put = ak4xxx_switch_put;
++			knew.access = 0;
++			/* register 1, bit 0 (SMUTE): 0 = normal operation,
++			   1 = mute */
++			knew.private_value =
++				AK_COMPOSE(idx/2, 1, 0, 0) | AK_INVERT;
++			err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
++			if (err < 0)
++				return err;
++		}
+ 		memset(&knew, 0, sizeof(knew));
+ 		if (! ak->dac_info || ! ak->dac_info[mixer_ch].name) {
+ 			knew.name = "DAC Volume";
+--- linux-2.6.22.1.orig/sound/isa/Kconfig
++++ linux-2.6.22.1/sound/isa/Kconfig
+@@ -1,8 +1,5 @@
+ # ALSA ISA drivers
+ 
+-menu "ISA devices"
+-	depends on SND!=n && ISA && ISA_DMA_API
+-
+ config SND_AD1848_LIB
+         tristate
+         select SND_PCM
+@@ -11,6 +8,22 @@
+         tristate
+         select SND_PCM
+ 
++config SND_SB_COMMON
++        tristate
++
++config SND_SB8_DSP
++        tristate
++        select SND_PCM
++        select SND_SB_COMMON
++
++config SND_SB16_DSP
++        tristate
++        select SND_PCM
++        select SND_SB_COMMON
++
++menu "ISA devices"
++	depends on SND!=n && ISA && ISA_DMA_API
++
+ config SND_ADLIB
+ 	tristate "AdLib FM card"
+ 	depends on SND
+@@ -55,7 +68,7 @@
+ 	select ISAPNP
+ 	select SND_OPL3_LIB
+ 	select SND_MPU401_UART
+-	select SND_PCM
++	select SND_SB16_DSP
+ 	help
+ 	  Say Y here to include support for soundcards based on Avance
+ 	  Logic ALS100, ALS110, ALS120 and ALS200 chips.
+@@ -81,6 +94,7 @@
+ 	tristate "C-Media CMI8330"
+ 	depends on SND
+ 	select SND_AD1848_LIB
++	select SND_SB16_DSP
+ 	help
+ 	  Say Y here to include support for soundcards based on the
+ 	  C-Media CMI8330 chip.
+@@ -132,7 +146,7 @@
+ 	select ISAPNP
+ 	select SND_OPL3_LIB
+ 	select SND_MPU401_UART
+-	select SND_PCM
++	select SND_SB16_DSP
+ 	help
+ 	  Say Y here to include support for soundcards based on the
+ 	  Diamond Technologies DT-019X or Avance Logic ALS-007 chips.
+@@ -145,7 +159,7 @@
+ 	depends on SND && PNP && ISA
+ 	select ISAPNP
+ 	select SND_MPU401_UART
+-	select SND_PCM
++	select SND_SB8_DSP
+ 	help
+ 	  Say Y here to include support for ESS AudioDrive ES968 chips.
+ 
+@@ -321,7 +335,7 @@
+ 	depends on SND
+ 	select SND_OPL3_LIB
+ 	select SND_RAWMIDI
+-	select SND_PCM
++	select SND_SB8_DSP
+ 	help
+ 	  Say Y here to include support for Creative Sound Blaster 1.0/
+ 	  2.0/Pro (8-bit) or 100% compatible soundcards.
+@@ -334,7 +348,7 @@
+ 	depends on SND
+ 	select SND_OPL3_LIB
+ 	select SND_MPU401_UART
+-	select SND_PCM
++	select SND_SB16_DSP
+ 	help
+ 	  Say Y here to include support for Sound Blaster 16 soundcards
+ 	  (including the Plug and Play version).
+@@ -347,7 +361,7 @@
+ 	depends on SND
+ 	select SND_OPL3_LIB
+ 	select SND_MPU401_UART
+-	select SND_PCM
++	select SND_SB16_DSP
+ 	help
+ 	  Say Y here to include support for Sound Blaster AWE soundcards
+ 	  (including the Plug and Play version).
+--- linux-2.6.22.1.orig/sound/isa/ad1848/ad1848_lib.c
++++ linux-2.6.22.1/sound/isa/ad1848/ad1848_lib.c
+@@ -245,7 +245,7 @@
+ 			snd_printk(KERN_ERR "mce_down - auto calibration time out (2)\n");
+ 			return;
+ 		}
+-		time = schedule_timeout_interruptible(time);
++		time = schedule_timeout(time);
+ 		spin_lock_irqsave(&chip->reg_lock, flags);
+ 	}
+ #if 0
+@@ -258,7 +258,7 @@
+ 			snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n");
+ 			return;
+ 		}
+-		time = schedule_timeout_interruptible(time);
++		time = schedule_timeout(time);
+ 		spin_lock_irqsave(&chip->reg_lock, flags);
+ 	}
+ 	spin_unlock_irqrestore(&chip->reg_lock, flags);
+--- linux-2.6.22.1.orig/sound/isa/opl3sa2.c
++++ linux-2.6.22.1/sound/isa/opl3sa2.c
+@@ -164,6 +164,8 @@
+ 	{ .id = "YMH0801", .devs = { { "YMH0021" } } },
+ 	/* NeoMagic MagicWave 3DX */
+ 	{ .id = "NMX2200", .devs = { { "YMH2210" } } },
++	/* NeoMagic MagicWave 3D */
++	{ .id = "NMX2200", .devs = { { "NMX2210" } } },
+ 	/* --- */
+ 	{ .id = "" }	/* end */
+ };
+--- linux-2.6.22.1.orig/sound/isa/opti9xx/opti92x-ad1848.c
++++ linux-2.6.22.1/sound/isa/opti9xx/opti92x-ad1848.c
+@@ -1927,10 +1927,12 @@
+ static int __devinit snd_opti9xx_isa_match(struct device *devptr,
+ 					   unsigned int dev)
+ {
++#ifdef CONFIG_PNP
+ 	if (snd_opti9xx_pnp_is_probed)
+ 		return 0;
+ 	if (isapnp)
+ 		return 0;
++#endif
+ 	return 1;
+ }
+ 
+@@ -2096,6 +2098,7 @@
+ 	pnp_register_card_driver(&opti9xx_pnpc_driver);
+ 	if (snd_opti9xx_pnp_is_probed)
+ 		return 0;
++	pnp_unregister_card_driver(&opti9xx_pnpc_driver);
+ #endif
+ 	return isa_register_driver(&snd_opti9xx_driver, 1);
+ }
+--- linux-2.6.22.1.orig/sound/isa/sb/Makefile
++++ linux-2.6.22.1/sound/isa/sb/Makefile
+@@ -22,14 +22,13 @@
+ sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+ 
+ # Toplevel Module Dependency
+-obj-$(CONFIG_SND_ALS100) += snd-sb16-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_CMI8330) += snd-sb16-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_DT019X) += snd-sb16-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_SB8) += snd-sb8.o snd-sb8-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_SB16) += snd-sb16.o snd-sb16-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o
+-obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o
++obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o
++obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o
++obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o
++obj-$(CONFIG_SND_SB8) += snd-sb8.o
++obj-$(CONFIG_SND_SB16) += snd-sb16.o
++obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o
++obj-$(CONFIG_SND_ES968) += snd-es968.o
+ ifeq ($(CONFIG_SND_SB16_CSP),y)
+   obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o
+   obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o
+--- linux-2.6.22.1.orig/sound/isa/sb/sb16_main.c
++++ linux-2.6.22.1/sound/isa/sb/sb16_main.c
+@@ -563,6 +563,11 @@
+       __open_ok:
+ 	if (chip->hardware == SB_HW_ALS100)
+ 		runtime->hw.rate_max = 48000;
++	if (chip->hardware == SB_HW_CS5530) {
++		runtime->hw.buffer_bytes_max = 32 * 1024;
++		runtime->hw.periods_min = 2;
++		runtime->hw.rate_min = 44100;
++	}
+ 	if (chip->mode & SB_RATE_LOCK)
+ 		runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate;
+ 	chip->playback_substream = substream;
+@@ -633,6 +638,11 @@
+       __open_ok:
+ 	if (chip->hardware == SB_HW_ALS100)
+ 		runtime->hw.rate_max = 48000;
++	if (chip->hardware == SB_HW_CS5530) {
++		runtime->hw.buffer_bytes_max = 32 * 1024;
++		runtime->hw.periods_min = 2;
++		runtime->hw.rate_min = 44100;
++	}
+ 	if (chip->mode & SB_RATE_LOCK)
+ 		runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate;
+ 	chip->capture_substream = substream;
+--- linux-2.6.22.1.orig/sound/isa/sb/sb_common.c
++++ linux-2.6.22.1/sound/isa/sb/sb_common.c
+@@ -128,7 +128,7 @@
+ 	minor = version & 0xff;
+ 	snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n",
+ 		    chip->port, major, minor);
+-	
++
+ 	switch (chip->hardware) {
+ 	case SB_HW_AUTO:
+ 		switch (major) {
+@@ -168,6 +168,9 @@
+ 	case SB_HW_DT019X:
+ 		str = "(DT019X/ALS007)";
+ 		break;
++	case SB_HW_CS5530:
++		str = "16 (CS5530)";
++		break;
+ 	default:
+ 		return -ENODEV;
+ 	}
+--- linux-2.6.22.1.orig/sound/isa/sb/sb_mixer.c
++++ linux-2.6.22.1/sound/isa/sb/sb_mixer.c
+@@ -821,6 +821,7 @@
+ 		break;
+ 	case SB_HW_16:
+ 	case SB_HW_ALS100:
++	case SB_HW_CS5530:
+ 		if ((err = snd_sbmixer_init(chip,
+ 					    snd_sb16_controls,
+ 					    ARRAY_SIZE(snd_sb16_controls),
+@@ -950,6 +951,7 @@
+ 		break;
+ 	case SB_HW_16:
+ 	case SB_HW_ALS100:
++	case SB_HW_CS5530:
+ 		save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs));
+ 		break;
+ 	case SB_HW_ALS4000:
+@@ -975,6 +977,7 @@
+ 		break;
+ 	case SB_HW_16:
+ 	case SB_HW_ALS100:
++	case SB_HW_CS5530:
+ 		restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs));
+ 		break;
+ 	case SB_HW_ALS4000:
+--- linux-2.6.22.1.orig/sound/isa/sscape.c
++++ linux-2.6.22.1/sound/isa/sscape.c
+@@ -382,7 +382,7 @@
+ 		unsigned long flags;
+ 		unsigned char x;
+ 
+-		schedule_timeout_interruptible(1);
++		schedule_timeout(1);
+ 
+ 		spin_lock_irqsave(&s->lock, flags);
+ 		x = inb(HOST_DATA_IO(s->io_base));
+@@ -409,7 +409,7 @@
+ 		unsigned long flags;
+ 		unsigned char x;
+ 
+-		schedule_timeout_interruptible(1);
++		schedule_timeout(1);
+ 
+ 		spin_lock_irqsave(&s->lock, flags);
+ 		x = inb(HOST_DATA_IO(s->io_base));
+--- linux-2.6.22.1.orig/sound/isa/wavefront/wavefront_synth.c
++++ linux-2.6.22.1/sound/isa/wavefront/wavefront_synth.c
+@@ -1780,7 +1780,7 @@
+ 	outb (val,port);
+ 	spin_unlock_irq(&dev->irq_lock);
+ 	while (1) {
+-		if ((timeout = schedule_timeout_interruptible(timeout)) == 0)
++		if ((timeout = schedule_timeout(timeout)) == 0)
+ 			return;
+ 		if (dev->irq_ok)
+ 			return;
+--- linux-2.6.22.1.orig/sound/pci/Kconfig
++++ linux-2.6.22.1/sound/pci/Kconfig
+@@ -33,6 +33,7 @@
+ 	select SND_OPL3_LIB
+ 	select SND_MPU401_UART
+ 	select SND_PCM
++	select SND_SB_COMMON
+ 	help
+ 	  Say Y here to include support for soundcards based on Avance Logic
+ 	  ALS4000 chips.
+@@ -215,6 +216,16 @@
+ 
+ 	  This works better than the old code, so say Y.
+ 
++config SND_CS5530
++	tristate "CS5530 Audio"
++	depends on SND && ISA_DMA_API
++	select SND_SB16_DSP
++	help
++	  Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips.
++
++	  To compile this driver as a module, choose M here: the module
++	  will be called snd-cs5530.
++
+ config SND_CS5535AUDIO
+ 	tristate "CS5535/CS5536 Audio"
+ 	depends on SND && X86 && !X86_64
+--- linux-2.6.22.1.orig/sound/pci/Makefile
++++ linux-2.6.22.1/sound/pci/Makefile
+@@ -12,6 +12,7 @@
+ snd-bt87x-objs := bt87x.o
+ snd-cmipci-objs := cmipci.o
+ snd-cs4281-objs := cs4281.o
++snd-cs5530-objs := cs5530.o
+ snd-ens1370-objs := ens1370.o
+ snd-ens1371-objs := ens1371.o
+ snd-es1938-objs := es1938.o
+@@ -36,6 +37,7 @@
+ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o
+ obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o
+ obj-$(CONFIG_SND_CS4281) += snd-cs4281.o
++obj-$(CONFIG_SND_CS5530) += snd-cs5530.o
+ obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o
+ obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o
+ obj-$(CONFIG_SND_ES1938) += snd-es1938.o
+--- linux-2.6.22.1.orig/sound/pci/ali5451/ali5451.c
++++ linux-2.6.22.1/sound/pci/ali5451/ali5451.c
+@@ -239,7 +239,7 @@
+ 
+ 
+ struct snd_ali {
+-	unsigned long	irq;
++	int		irq;
+ 	unsigned long	port;
+ 	unsigned char	revision;
+ 
+@@ -731,8 +731,7 @@
+ 		return;
+ 	}
+ 
+-	count = 0;
+-	while (count++ <= 50000) {
++	for (count = 0; count <= 50000; count++) {
+ 		snd_ali_delay(codec, 6);
+ 		bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1));
+ 		R2 = bval & 0x1F;
+@@ -2343,7 +2342,7 @@
+ 	strcpy(card->driver, "ALI5451");
+ 	strcpy(card->shortname, "ALI 5451");
+ 	
+-	sprintf(card->longname, "%s at 0x%lx, irq %li",
++	sprintf(card->longname, "%s at 0x%lx, irq %i",
+ 		card->shortname, codec->port, codec->irq);
+ 
+ 	snd_ali_printk("register card.\n");
+--- linux-2.6.22.1.orig/sound/pci/als300.c
++++ linux-2.6.22.1/sound/pci/als300.c
+@@ -88,8 +88,8 @@
+ #define PLAYBACK_BLOCK_COUNTER	0x9A
+ #define RECORD_BLOCK_COUNTER	0x9B
+ 
+-#define DEBUG_CALLS	1
+-#define DEBUG_PLAY_REC	1
++#define DEBUG_CALLS	0
++#define DEBUG_PLAY_REC	0
+ 
+ #if DEBUG_CALLS
+ #define snd_als300_dbgcalls(format, args...) printk(format, ##args)
+@@ -733,7 +733,8 @@
+ 
+ 	snd_als300_init(chip);
+ 
+-	if (snd_als300_ac97(chip) < 0) {
++	err = snd_als300_ac97(chip);
++	if (err < 0) {
+ 		snd_printk(KERN_WARNING "Could not create ac97\n");
+ 		snd_als300_free(chip);
+ 		return err;
+--- linux-2.6.22.1.orig/sound/pci/ca0106/ca0106_main.c
++++ linux-2.6.22.1/sound/pci/ca0106/ca0106_main.c
+@@ -168,6 +168,25 @@
+ #include "ca0106.h"
+ 
+ static struct snd_ca0106_details ca0106_chip_details[] = {
++	 /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
++	 /* It is really just a normal SB Live 24bit. */
++	 /*
++ 	  * CTRL:CA0111-WTLF
++	  * ADC: WM8775SEDS
++	  * DAC: CS4382-KQZ
++	  */
++	 /* Tested:
++	  * Playback on front, rear, center/lfe speakers
++	  * Capture from Mic in.
++	  * Not-Tested:
++	  * Capture from Line in.
++	  * Playback to digital out.
++	  */
++	 { .serial = 0x10121102,
++	   .name   = "X-Fi Extreme Audio [SB0790]",
++	   .gpio_type = 1,
++	   .i2c_adc = 1 } ,
++	 /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97.  */
+ 	 /* AudigyLS[SB0310] */
+ 	 { .serial = 0x10021102,
+ 	   .name   = "AudigyLS [SB0310]",
+--- linux-2.6.22.1.orig/sound/pci/cs46xx/cs46xx_lib.c
++++ linux-2.6.22.1/sound/pci/cs46xx/cs46xx_lib.c
+@@ -2897,6 +2897,10 @@
+ 	}
+ #endif
+ 	
++#ifdef CONFIG_PM
++	kfree(chip->saved_regs);
++#endif
++
+ 	pci_disable_device(chip->pci);
+ 	kfree(chip);
+ 	return 0;
+@@ -3140,6 +3144,23 @@
+ /*
+  *  start and load DSP 
+  */
++
++static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip)
++{
++	unsigned int tmp;
++
++	snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM);
++        
++	tmp = snd_cs46xx_peek(chip, BA1_PFIE);
++	tmp &= ~0x0000f03f;
++	snd_cs46xx_poke(chip, BA1_PFIE, tmp);	/* playback interrupt enable */
++
++	tmp = snd_cs46xx_peek(chip, BA1_CIE);
++	tmp &= ~0x0000003f;
++	tmp |=  0x00000001;
++	snd_cs46xx_poke(chip, BA1_CIE, tmp);	/* capture interrupt enable */
++}
++
+ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip)
+ {	
+ 	unsigned int tmp;
+@@ -3214,19 +3235,7 @@
+ 
+ 	snd_cs46xx_proc_start(chip);
+ 
+-	/*
+-	 *  Enable interrupts on the part.
+-	 */
+-	snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM);
+-        
+-	tmp = snd_cs46xx_peek(chip, BA1_PFIE);
+-	tmp &= ~0x0000f03f;
+-	snd_cs46xx_poke(chip, BA1_PFIE, tmp);	/* playback interrupt enable */
+-
+-	tmp = snd_cs46xx_peek(chip, BA1_CIE);
+-	tmp &= ~0x0000003f;
+-	tmp |=  0x00000001;
+-	snd_cs46xx_poke(chip, BA1_CIE, tmp);	/* capture interrupt enable */
++	cs46xx_enable_stream_irqs(chip);
+ 	
+ #ifndef CONFIG_SND_CS46XX_NEW_DSP
+ 	/* set the attenuation to 0dB */ 
+@@ -3665,11 +3674,19 @@
+  * APM support
+  */
+ #ifdef CONFIG_PM
++static unsigned int saved_regs[] = {
++	BA0_ACOSV,
++	BA0_ASER_FADDR,
++	BA0_ASER_MASTER,
++	BA1_PVOL,
++	BA1_CVOL,
++};
++
+ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
+ {
+ 	struct snd_card *card = pci_get_drvdata(pci);
+ 	struct snd_cs46xx *chip = card->private_data;
+-	int amp_saved;
++	int i, amp_saved;
+ 
+ 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ 	chip->in_suspend = 1;
+@@ -3680,6 +3697,10 @@
+ 	snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]);
+ 	snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]);
+ 
++	/* save some registers */
++	for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
++		chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]);
++
+ 	amp_saved = chip->amplifier;
+ 	/* turn off amp */
+ 	chip->amplifier_ctrl(chip, -chip->amplifier);
+@@ -3698,7 +3719,7 @@
+ {
+ 	struct snd_card *card = pci_get_drvdata(pci);
+ 	struct snd_cs46xx *chip = card->private_data;
+-	int amp_saved;
++	int i, amp_saved;
+ 
+ 	pci_set_power_state(pci, PCI_D0);
+ 	pci_restore_state(pci);
+@@ -3716,6 +3737,16 @@
+ 
+ 	snd_cs46xx_chip_init(chip);
+ 
++	snd_cs46xx_reset(chip);
++#ifdef CONFIG_SND_CS46XX_NEW_DSP
++	cs46xx_dsp_resume(chip);
++	/* restore some registers */
++	for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
++		snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]);
++#else
++	snd_cs46xx_download_image(chip);
++#endif
++
+ #if 0
+ 	snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, 
+ 			       chip->ac97_general_purpose);
+@@ -3730,6 +3761,13 @@
+ 	snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]);
+ 	snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]);
+ 
++	/* reset playback/capture */
++	snd_cs46xx_set_play_sample_rate(chip, 8000);
++	snd_cs46xx_set_capture_sample_rate(chip, 8000);
++	snd_cs46xx_proc_start(chip);
++
++	cs46xx_enable_stream_irqs(chip);
++
+ 	if (amp_saved)
+ 		chip->amplifier_ctrl(chip, 1); /* turn amp on */
+ 	else
+@@ -3896,6 +3934,15 @@
+ 	
+ 	snd_cs46xx_proc_init(card, chip);
+ 
++#ifdef CONFIG_PM
++	chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) *
++				   ARRAY_SIZE(saved_regs), GFP_KERNEL);
++	if (!chip->saved_regs) {
++		snd_cs46xx_free(chip);
++		return -ENOMEM;
++	}
++#endif
++
+ 	chip->active_ctrl(chip, -1); /* disable CLKRUN */
+ 
+ 	snd_card_set_dev(card, &pci->dev);
+--- linux-2.6.22.1.orig/sound/pci/cs46xx/cs46xx_lib.h
++++ linux-2.6.22.1/sound/pci/cs46xx/cs46xx_lib.h
+@@ -86,6 +86,9 @@
+ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip);
+ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip);
+ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module);
++#ifdef CONFIG_PM
++int cs46xx_dsp_resume(struct snd_cs46xx * chip);
++#endif
+ struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name,
+ 						   int symbol_type);
+ #ifdef CONFIG_PROC_FS
+--- linux-2.6.22.1.orig/sound/pci/cs46xx/dsp_spos.c
++++ linux-2.6.22.1/sound/pci/cs46xx/dsp_spos.c
+@@ -306,13 +306,59 @@
+ 	mutex_unlock(&chip->spos_mutex);
+ }
+ 
++static int dsp_load_parameter(struct snd_cs46xx *chip,
++			      struct dsp_segment_desc *parameter)
++{
++	u32 doffset, dsize;
++
++	if (!parameter) {
++		snd_printdd("dsp_spos: module got no parameter segment\n");
++		return 0;
++	}
++
++	doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET);
++	dsize   = parameter->size * 4;
++
++	snd_printdd("dsp_spos: "
++		    "downloading parameter data to chip (%08x-%08x)\n",
++		    doffset,doffset + dsize);
++	if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) {
++		snd_printk(KERN_ERR "dsp_spos: "
++			   "failed to download parameter data to DSP\n");
++		return -EINVAL;
++	}
++	return 0;
++}
++
++static int dsp_load_sample(struct snd_cs46xx *chip,
++			   struct dsp_segment_desc *sample)
++{
++	u32 doffset, dsize;
++
++	if (!sample) {
++		snd_printdd("dsp_spos: module got no sample segment\n");
++		return 0;
++	}
++
++	doffset = (sample->offset * 4  + DSP_SAMPLE_BYTE_OFFSET);
++	dsize   =  sample->size * 4;
++
++	snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n",
++		    doffset,doffset + dsize);
++
++	if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) {
++		snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n");
++		return -EINVAL;
++	}
++	return 0;
++}
++
+ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module)
+ {
+ 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ 	struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM);
+-	struct dsp_segment_desc * parameter = get_segment_desc (module,SEGTYPE_SP_PARAMETER);
+-	struct dsp_segment_desc * sample = get_segment_desc (module,SEGTYPE_SP_SAMPLE);
+ 	u32 doffset, dsize;
++	int err;
+ 
+ 	if (ins->nmodules == DSP_MAX_MODULES - 1) {
+ 		snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n");
+@@ -326,49 +372,20 @@
+ 		snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE);
+ 	}
+   
+-	if (parameter == NULL) {
+-		snd_printdd("dsp_spos: module got no parameter segment\n");
+-	} else {
+-		if (ins->nmodules > 0) {
+-			snd_printk(KERN_WARNING "dsp_spos: WARNING current parameter data may be overwriten!\n");
+-		}
+-
+-		doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET);
+-		dsize   = parameter->size * 4;
+-
+-		snd_printdd("dsp_spos: downloading parameter data to chip (%08x-%08x)\n",
+-			    doffset,doffset + dsize);
+-
+-		if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) {
+-			snd_printk(KERN_ERR "dsp_spos: failed to download parameter data to DSP\n");
+-			return -EINVAL;
+-		}
+-	}
++	err = dsp_load_parameter(chip, get_segment_desc(module,
++							SEGTYPE_SP_PARAMETER));
++	if (err < 0)
++		return err;
+ 
+ 	if (ins->nmodules == 0) {
+ 		snd_printdd("dsp_spos: clearing sample area\n");
+ 		snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE);
+ 	}
+ 
+-	if (sample == NULL) {
+-		snd_printdd("dsp_spos: module got no sample segment\n");
+-	} else {
+-		if (ins->nmodules > 0) {
+-			snd_printk(KERN_WARNING "dsp_spos: WARNING current sample data may be overwriten\n");
+-		}
+-
+-		doffset = (sample->offset * 4  + DSP_SAMPLE_BYTE_OFFSET);
+-		dsize   =  sample->size * 4;
+-
+-		snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n",
+-			    doffset,doffset + dsize);
+-
+-		if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) {
+-			snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n");
+-			return -EINVAL;
+-		}
+-	}
+-
++	err = dsp_load_sample(chip, get_segment_desc(module,
++						     SEGTYPE_SP_SAMPLE));
++	if (err < 0)
++		return err;
+ 
+ 	if (ins->nmodules == 0) {
+ 		snd_printdd("dsp_spos: clearing code area\n");
+@@ -986,7 +1003,10 @@
+ 		return NULL;
+ 	}
+ 
+-	strcpy(ins->tasks[ins->ntask].task_name,name);
++	if (name)
++		strcpy(ins->tasks[ins->ntask].task_name, name);
++	else
++		strcpy(ins->tasks[ins->ntask].task_name, "(NULL)");
+ 	ins->tasks[ins->ntask].address = dest;
+ 	ins->tasks[ins->ntask].size = size;
+ 
+@@ -995,7 +1015,8 @@
+ 	desc = (ins->tasks + ins->ntask);
+ 	ins->ntask++;
+ 
+-	add_symbol (chip,name,dest,SYMBOL_PARAMETER);
++	if (name)
++		add_symbol (chip,name,dest,SYMBOL_PARAMETER);
+ 	return desc;
+ }
+ 
+@@ -1006,6 +1027,7 @@
+ 
+ 	desc = _map_scb (chip,name,dest);
+ 	if (desc) {
++		desc->data = scb_data;
+ 		_dsp_create_scb(chip,scb_data,dest);
+ 	} else {
+ 		snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n");
+@@ -1023,6 +1045,7 @@
+ 
+ 	desc = _map_task_tree (chip,name,dest,size);
+ 	if (desc) {
++		desc->data = task_data;
+ 		_dsp_create_task_tree(chip,task_data,dest,size);
+ 	} else {
+ 		snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n");
+@@ -1320,8 +1343,10 @@
+ 			0x0000ffff
+ 		};
+     
+-		/* dirty hack ... */
+-		_dsp_create_task_tree (chip,(u32 *)&mix2_ostream_spb,WRITE_BACK_SPB,2);
++		if (!cs46xx_dsp_create_task_tree(chip, NULL,
++						 (u32 *)&mix2_ostream_spb,
++						 WRITE_BACK_SPB, 2))
++			goto _fail_end;
+ 	}
+ 
+ 	/* input sample converter */
+@@ -1622,7 +1647,6 @@
+ 	return 0;
+ }
+ 
+-
+ static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip)
+ {
+ 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+@@ -1894,3 +1918,61 @@
+ 
+ 	return 0;
+ }
++
++#ifdef CONFIG_PM
++int cs46xx_dsp_resume(struct snd_cs46xx * chip)
++{
++	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
++	int i, err;
++
++	/* clear parameter, sample and code areas */
++	snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET,
++			     DSP_PARAMETER_BYTE_SIZE);
++	snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET,
++			     DSP_SAMPLE_BYTE_SIZE);
++	snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE);
++
++	for (i = 0; i < ins->nmodules; i++) {
++		struct dsp_module_desc *module = &ins->modules[i];
++		struct dsp_segment_desc *seg;
++		u32 doffset, dsize;
++
++		seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER);
++		err = dsp_load_parameter(chip, seg);
++		if (err < 0)
++			return err;
++
++		seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE);
++		err = dsp_load_sample(chip, seg);
++		if (err < 0)
++			return err;
++
++		seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM);
++		if (!seg)
++			continue;
++
++		doffset = seg->offset * 4 + module->load_address * 4
++			+ DSP_CODE_BYTE_OFFSET;
++		dsize   = seg->size * 4;
++		err = snd_cs46xx_download(chip,
++					  ins->code.data + module->load_address,
++					  doffset, dsize);
++		if (err < 0)
++			return err;
++	}
++
++	for (i = 0; i < ins->ntask; i++) {
++		struct dsp_task_descriptor *t = &ins->tasks[i];
++		_dsp_create_task_tree(chip, t->data, t->address, t->size);
++	}
++
++	for (i = 0; i < ins->nscb; i++) {
++		struct dsp_scb_descriptor *s = &ins->scbs[i];
++		if (s->deleted)
++			continue;
++		_dsp_create_scb(chip, s->data, s->address);
++	}
++
++	return 0;
++}
++#endif
+--- /dev/null
++++ linux-2.6.22.1/sound/pci/cs5530.c
+@@ -0,0 +1,306 @@
++/*
++ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
++ *
++ * 	(C) Copyright 2007 Ash Willis <ashwillis at programmer.net>
++ *	(C) Copyright 2003 Red Hat Inc <alan at redhat.com>
++ *
++ * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
++ * mess with it a bit. The chip seems to have to have trouble with full duplex
++ * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to
++ * simultaneously play back audio at 16bit 44100kHz, the device actually plays
++ * back in the same format in which it is capturing. By forcing the chip to
++ * always play/capture in 16/44100, we can let alsa-lib convert the samples and
++ * that way we can hack up some full duplex audio. 
++ * 
++ * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
++ * The older version (VSA1) provides fairly good soundblaster emulation
++ * although there are a couple of bugs: large DMA buffers break record,
++ * and the MPU event handling seems suspect. VSA2 allows the native driver
++ * to control the AC97 audio engine directly and requires a different driver.
++ *
++ * Thanks to National Semiconductor for providing the needed information
++ * on the XpressAudio(tm) internals.
++ *
++ * This program is free software; you can redistribute it and/or modify it
++ * under the terms of the GNU General Public License as published by the
++ * Free Software Foundation; either version 2, or (at your option) any
++ * later version.
++ *
++ * This program is distributed in the hope that it will be useful, but
++ * WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
++ * General Public License for more details.
++ *
++ * TO DO:
++ *	Investigate whether we can portably support Cognac (5520) in the
++ *	same manner.
++ */
++
++#include <sound/driver.h>
++#include <linux/delay.h>
++#include <linux/moduleparam.h>
++#include <linux/pci.h>
++#include <sound/core.h>
++#include <sound/sb.h>
++#include <sound/initval.h>
++
++MODULE_AUTHOR("Ash Willis");
++MODULE_DESCRIPTION("CS5530 Audio");
++MODULE_LICENSE("GPL");
++
++static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
++static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
++static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
++
++struct snd_cs5530 {
++	struct snd_card *card;
++	struct pci_dev *pci;
++	struct snd_sb *sb;
++	unsigned long pci_base;
++};
++
++static struct pci_device_id snd_cs5530_ids[] = {
++	{PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID,
++							PCI_ANY_ID, 0, 0},
++	{0,}
++};
++
++MODULE_DEVICE_TABLE(pci, snd_cs5530_ids);
++
++static int snd_cs5530_free(struct snd_cs5530 *chip)
++{
++	pci_release_regions(chip->pci);
++	pci_disable_device(chip->pci);
++	kfree(chip);
++	return 0;
++}
++
++static int snd_cs5530_dev_free(struct snd_device *device)
++{
++	struct snd_cs5530 *chip = device->device_data;
++	return snd_cs5530_free(chip);
++}
++
++static void __devexit snd_cs5530_remove(struct pci_dev *pci)
++{
++	snd_card_free(pci_get_drvdata(pci));
++	pci_set_drvdata(pci, NULL);
++}
++
++static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg)
++{
++	outb(reg, io + 4);
++	udelay(20);
++	reg = inb(io + 5);
++	udelay(20);
++	return reg;
++}
++
++static int __devinit snd_cs5530_create(struct snd_card *card,
++				       struct pci_dev *pci,
++				       struct snd_cs5530 **rchip)
++{
++	struct snd_cs5530 *chip;
++	unsigned long sb_base;
++	u8 irq, dma8, dma16 = 0;
++	u16 map;
++	void __iomem *mem;
++	int err;
++
++	static struct snd_device_ops ops = {
++		.dev_free = snd_cs5530_dev_free,
++	};
++	*rchip = NULL;
++
++	err = pci_enable_device(pci);
++ 	if (err < 0)
++		return err;
++
++	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
++	if (chip == NULL) {
++		pci_disable_device(pci);
++		return -ENOMEM;
++	}
++
++	chip->card = card;
++	chip->pci = pci;
++
++	err = pci_request_regions(pci, "CS5530");
++	if (err < 0) {
++		kfree(chip); 
++		pci_disable_device(pci);
++		return err;
++	}
++	chip->pci_base = pci_resource_start(pci, 0);
++
++	mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0));
++	if (mem == NULL) {
++		kfree(chip);
++		pci_disable_device(pci);
++		return -EBUSY;
++	}
++
++	map = readw(mem + 0x18);
++	iounmap(mem);
++
++	/* Map bits
++		0:1	* 0x20 + 0x200 = sb base
++		2	sb enable
++		3	adlib enable
++		5	MPU enable 0x330
++		6	MPU enable 0x300
++
++	   The other bits may be used internally so must be masked */
++
++	sb_base = 0x220 + 0x20 * (map & 3);
++
++	if (map & (1<<2))
++		printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base);
++	else {
++		printk(KERN_ERR "Could not find XpressAudio!\n");
++		snd_cs5530_free(chip);
++		return -ENODEV;
++	}
++
++	if (map & (1<<5))
++		printk(KERN_INFO "CS5530: MPU at 0x300\n");
++	else if (map & (1<<6))
++		printk(KERN_INFO "CS5530: MPU at 0x330\n");
++
++	irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F;
++	dma8 = snd_cs5530_mixer_read(sb_base, 0x81);
++
++	if (dma8 & 0x20)
++		dma16 = 5;
++	else if (dma8 & 0x40)
++		dma16 = 6;
++	else if (dma8 & 0x80)
++		dma16 = 7;
++	else {
++		printk(KERN_ERR "CS5530: No 16bit DMA enabled\n");
++		snd_cs5530_free(chip);
++		return -ENODEV;
++	}
++
++	if (dma8 & 0x01)
++		dma8 = 0;
++	else if (dma8 & 02)
++		dma8 = 1;
++	else if (dma8 & 0x08)
++		dma8 = 3;
++	else {
++		printk(KERN_ERR "CS5530: No 8bit DMA enabled\n");
++		snd_cs5530_free(chip);
++		return -ENODEV;
++	}
++
++	if (irq & 1)
++		irq = 9;
++	else if (irq & 2)
++		irq = 5;
++	else if (irq & 4)
++		irq = 7;
++	else if (irq & 8)
++		irq = 10;
++	else {
++		printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n");
++		snd_cs5530_free(chip);
++		return -ENODEV;
++	}
++
++	printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, 
++									dma16);
++
++	err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8,
++						dma16, SB_HW_CS5530, &chip->sb);
++	if (err < 0) {
++		printk(KERN_ERR "CS5530: Could not create SoundBlaster\n");
++		snd_cs5530_free(chip);
++		return err;
++	}
++
++	err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm);
++	if (err < 0) {
++		printk(KERN_ERR "CS5530: Could not create PCM\n");
++		snd_cs5530_free(chip);
++		return err;
++	}
++
++	err = snd_sbmixer_new(chip->sb);
++	if (err < 0) {
++		printk(KERN_ERR "CS5530: Could not create Mixer\n");
++		snd_cs5530_free(chip);
++		return err;
++	}
++
++	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
++	if (err < 0) {
++		snd_cs5530_free(chip);
++		return err;
++	}
++
++	snd_card_set_dev(card, &pci->dev);
++	*rchip = chip;
++	return 0;
++}
++
++static int __devinit snd_cs5530_probe(struct pci_dev *pci,
++					const struct pci_device_id *pci_id)
++{
++	static int dev;
++	struct snd_card *card;
++	struct snd_cs5530 *chip = NULL;
++	int err;
++
++	if (dev >= SNDRV_CARDS)
++		return -ENODEV;
++	if (!enable[dev]) {
++		dev++;
++		return -ENOENT;
++	}
++
++	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
++
++	if (card == NULL)
++		return -ENOMEM;
++
++	err = snd_cs5530_create(card, pci, &chip);
++	if (err < 0) {
++		snd_card_free(card);
++		return err;
++	}
++
++	strcpy(card->driver, "CS5530");
++	strcpy(card->shortname, "CS5530 Audio");
++	sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base);
++
++	err = snd_card_register(card);
++	if (err < 0) {
++		snd_card_free(card);
++		return err;
++	}
++	pci_set_drvdata(pci, card);
++	dev++;
++	return 0;
++}
++
++static struct pci_driver driver = {
++	.name = "CS5530_Audio",
++	.id_table = snd_cs5530_ids,
++	.probe = snd_cs5530_probe,
++	.remove = __devexit_p(snd_cs5530_remove),
++};
++
++static int __init alsa_card_cs5530_init(void)
++{
++	return pci_register_driver(&driver);
++}
++
++static void __exit alsa_card_cs5530_exit(void)
++{
++	pci_unregister_driver(&driver);
++}
++
++module_init(alsa_card_cs5530_init)
++module_exit(alsa_card_cs5530_exit)
++
+--- linux-2.6.22.1.orig/sound/pci/emu10k1/emu10k1_main.c
++++ linux-2.6.22.1/sound/pci/emu10k1/emu10k1_main.c
+@@ -51,9 +51,15 @@
+ 
+ #define HANA_FILENAME "emu/hana.fw"
+ #define DOCK_FILENAME "emu/audio_dock.fw"
++#define EMU1010B_FILENAME "emu/emu1010b.fw"
++#define MICRO_DOCK_FILENAME "emu/micro_dock.fw"
++#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw"
+ 
+ MODULE_FIRMWARE(HANA_FILENAME);
+ MODULE_FIRMWARE(DOCK_FILENAME);
++MODULE_FIRMWARE(EMU1010B_FILENAME);
++MODULE_FIRMWARE(MICRO_DOCK_FILENAME);
++MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME);
+ 
+ 
+ /*************************************************************************
+@@ -660,10 +666,12 @@
+ 		return err;
+ 	}
+ 	snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size);
++#if 0
+ 	if (fw_entry->size != 0x133a4) {
+ 		snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename);
+ 		return -EINVAL;
+ 	}
++#endif
+ 
+ 	/* The FPGA is a Xilinx Spartan IIE XC2S50E */
+ 	/* GPIO7 -> FPGA PGMN
+@@ -694,6 +702,37 @@
+ 	return 0;
+ }
+ 
++/*
++ * EMU-1010 - details found out from this driver, official MS Win drivers,
++ * testing the card:
++ *
++ * Audigy2 (aka Alice2):
++ * ---------------------
++ * 	* communication over PCI
++ * 	* conversion of 32-bit data coming over EMU32 links from HANA FPGA
++ *	  to 2 x 16-bit, using internal DSP instructions
++ * 	* slave mode, clock supplied by HANA
++ * 	* linked to HANA using:
++ * 		32 x 32-bit serial EMU32 output channels
++ * 		16 x EMU32 input channels
++ * 		(?) x I2S I/O channels (?)
++ *
++ * FPGA (aka HANA):
++ * ---------------
++ * 	* provides all (?) physical inputs and outputs of the card
++ * 		(ADC, DAC, SPDIF I/O, ADAT I/O, etc.)
++ * 	* provides clock signal for the card and Alice2
++ * 	* two crystals - for 44.1kHz and 48kHz multiples
++ * 	* provides internal routing of signal sources to signal destinations
++ * 	* inputs/outputs to Alice2 - see above
++ *
++ * Current status of the driver:
++ * ----------------------------
++ * 	* only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz)
++ * 	* PCM device nb. 2:
++ *		16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops
++ * 		16 x 32-bit capture - snd_emu10k1_capture_efx_ops
++ */
+ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
+ {
+ 	unsigned int i;
+@@ -727,7 +766,7 @@
+ 	/* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */
+ 	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ 	snd_printdd("reg1=0x%x\n",reg);
+-	if (reg == 0x55) {
++	if ((reg & 0x3f) == 0x15) {
+ 		/* FPGA netlist already present so clear it */
+ 		/* Return to programming mode */
+ 
+@@ -735,19 +774,32 @@
+ 	}
+ 	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ 	snd_printdd("reg2=0x%x\n",reg);
+-	if (reg == 0x55) {
++	if ((reg & 0x3f) == 0x15) {
+ 		/* FPGA failed to return to programming mode */
++		snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n");
+ 		return -ENODEV;
+ 	}
+ 	snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg);
+-	if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) {
+-		snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME);
+-		return err;
++	if (emu->card_capabilities->emu1010 == 1) {
++		if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) {
++			snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME);
++			return err;
++		}
++	} else if (emu->card_capabilities->emu1010 == 2) {
++		if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) {
++			snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME);
++			return err;
++		}
++	} else if (emu->card_capabilities->emu1010 == 3) {
++		if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) {
++			snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME);
++			return err;
++		}
+ 	}
+ 
+ 	/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ 	snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+-	if (reg != 0x55) {
++	if ((reg & 0x3f) != 0x15) {
+ 		/* FPGA failed to be programmed */
+ 		snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg);
+ 		return -ENODEV;
+@@ -850,6 +902,27 @@
+ 		EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1);
+ 	snd_emu1010_fpga_link_dst_src_write(emu,
+ 		EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1);
++	/* Pavel Hofman - setting defaults for 8 more capture channels
++	 * Defaults only, users will set their own values anyways, let's
++	 * just copy/paste.
++	 */
++	
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1);
++	snd_emu1010_fpga_link_dst_src_write(emu,
++		EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1);
+ #endif
+ #if 0
+ 	/* Original */
+@@ -943,16 +1016,27 @@
+ 		/* Return to Audio Dock programming mode */
+ 		snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
+ 		snd_emu1010_fpga_write(emu,  EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
+-		if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+-			return err;
++		if (emu->card_capabilities->emu1010 == 1) {
++			if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
++				return err;
++			}
++		} else if (emu->card_capabilities->emu1010 == 2) {
++			if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
++				return err;
++			}
++		} else if (emu->card_capabilities->emu1010 == 3) {
++			if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
++				return err;
++			}
+ 		}
++
+ 		snd_emu1010_fpga_write(emu,  EMU_HANA_FPGA_CONFIG, 0 );
+ 		snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
+ 		snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
+ 		/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ 		snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ 		snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
+-		if (reg != 0x55) {
++		if ((reg & 0x3f) != 0x15) {
+ 			/* FPGA failed to be programmed */
+ 			snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
+ 			return 0;
+@@ -1227,9 +1311,15 @@
+ 	 .emu10k2_chip = 1,
+ 	 .ca0108_chip = 1,
+ 	 .ca_cardbus_chip = 1,
+-	 .spi_dac = 1,
+-	 .i2c_adc = 1,
+-	 .spk71 = 1} ,
++	 .spk71 = 1 ,
++	 .emu1010 = 3} ,
++	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102,
++	 .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", 
++	 .id = "EMU1010",
++	 .emu10k2_chip = 1,
++	 .ca0108_chip = 1,
++	 .spk71 = 1 ,
++	 .emu1010 = 2} ,
+ 	{.vendor = 0x1102, .device = 0x0008, 
+ 	 .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", 
+ 	 .id = "Audigy2",
+@@ -1665,12 +1755,13 @@
+ 	emu->fx8010.extout_mask = extout_mask;
+ 	emu->enable_ir = enable_ir;
+ 
++	if (emu->card_capabilities->ca_cardbus_chip) {
++		if ((err = snd_emu10k1_cardbus_init(emu)) < 0)
++			goto error;
++	}
+ 	if (emu->card_capabilities->ecard) {
+ 		if ((err = snd_emu10k1_ecard_init(emu)) < 0)
+ 			goto error;
+-	} else if (emu->card_capabilities->ca_cardbus_chip) {
+-		if ((err = snd_emu10k1_cardbus_init(emu)) < 0)
+-			goto error;
+  	} else if (emu->card_capabilities->emu1010) {
+  		if ((err = snd_emu10k1_emu1010_init(emu)) < 0) {
+  			snd_emu10k1_free(emu);
+@@ -1816,10 +1907,10 @@
+ 
+ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu)
+ {
++	if (emu->card_capabilities->ca_cardbus_chip)
++		snd_emu10k1_cardbus_init(emu);
+ 	if (emu->card_capabilities->ecard)
+ 		snd_emu10k1_ecard_init(emu);
+-	else if (emu->card_capabilities->ca_cardbus_chip)
+-		snd_emu10k1_cardbus_init(emu);
+ 	else if (emu->card_capabilities->emu1010)
+  		snd_emu10k1_emu1010_init(emu);
+ 	else
+--- linux-2.6.22.1.orig/sound/pci/emu10k1/emufx.c
++++ linux-2.6.22.1/sound/pci/emu10k1/emufx.c
+@@ -1123,6 +1123,11 @@
+ 	ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF;
+ }
+ 
++/*
++ * Used for emu1010 - conversion from 32-bit capture inputs from HANA
++ * to 2 x 16-bit registers in audigy - their values are read via DMA.
++ * Conversion is performed by Audigy DSP instructions of FX8010.
++ */
+ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16(
+ 				struct snd_emu10k1_fx8010_code *icode,
+ 				u32 *ptr, int tmp, int bit_shifter16,
+@@ -1193,7 +1198,11 @@
+ 	snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP);
+ 
+ #if 1
+-	/* PCM front Playback Volume (independent from stereo mix) */
++	/* PCM front Playback Volume (independent from stereo mix)
++	 * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31)
++	 * where gpr contains attenuation from corresponding mixer control
++	 * (snd_emu10k1_init_stereo_control)
++	 */
+ 	A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT));
+ 	A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT));
+ 	snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100);
+@@ -1549,7 +1558,7 @@
+ 
+ 	if (emu->card_capabilities->emu1010) {
+ 		snd_printk("EMU inputs on\n");
+-		/* Capture 8 channels of S32_LE sound */
++		/* Capture 16 (originally 8) channels of S32_LE sound */
+ 		
+ 		/* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+ 		/* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
+@@ -1560,6 +1569,11 @@
+ 		snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
+ 		/* Right ADC in 1 of 2 */
+ 		gpr_map[gpr++] = 0x00000000;
++		/* Delaying by one sample: instead of copying the input
++		 * value A_P16VIN to output A_FXBUS2 as in the first channel,
++		 * we use an auxiliary register, delaying the value by one
++		 * sample
++		 */
+ 		snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
+ 		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
+ 		gpr_map[gpr++] = 0x00000000;
+@@ -1583,6 +1597,66 @@
+ 		gpr_map[gpr++] = 0x00000000;
+ 		snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
+ 		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
++		/* Pavel Hofman - we still have voices, A_FXBUS2s, and
++		 * A_P16VINs available -
++		 * let's add 8 more capture channels - total of 16
++		 */
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x10));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x12));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x14));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x16));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x18));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x1a));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x1c));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
++		     A_C_00000000, A_C_00000000);
++		gpr_map[gpr++] = 0x00000000;
++		snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
++							  bit_shifter16,
++							  A_GPR(gpr - 1),
++							  A_FXBUS2(0x1e));
++		A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
++		     A_C_00000000, A_C_00000000);
+ 
+ #if 0
+ 		for (z = 4; z < 8; z++) {
+--- linux-2.6.22.1.orig/sound/pci/emu10k1/emumixer.c
++++ linux-2.6.22.1/sound/pci/emu10k1/emumixer.c
+@@ -77,6 +77,10 @@
+ 	return 0;
+ }
+ 
++/*
++ * Items labels in enum mixer controls assigning source data to
++ * each destination
++ */
+ static char *emu1010_src_texts[] = { 
+ 	"Silence",
+ 	"Dock Mic A",
+@@ -133,6 +137,9 @@
+ 	"DSP 31",
+ };
+ 
++/*
++ * List of data sources available for each destination
++ */
+ static unsigned int emu1010_src_regs[] = {
+ 	EMU_SRC_SILENCE,/* 0 */
+ 	EMU_SRC_DOCK_MIC_A1, /* 1 */
+@@ -189,6 +196,10 @@
+ 	EMU_SRC_ALICE_EMU32B+0xf, /* 52 */
+ };
+ 
++/*
++ * Data destinations - physical EMU outputs.
++ * Each destination has an enum mixer control to choose a data source
++ */
+ static unsigned int emu1010_output_dst[] = {
+ 	EMU_DST_DOCK_DAC1_LEFT1, /* 0 */
+ 	EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */
+@@ -216,6 +227,11 @@
+ 	EMU_DST_HANA_ADAT+7, /* 23 */
+ };
+ 
++/*
++ * Data destinations - HANA outputs going to Alice2 (audigy) for
++ *   capture (EMU32 + I2S links)
++ * Each destination has an enum mixer control to choose a data source
++ */
+ static unsigned int emu1010_input_dst[] = {
+ 	EMU_DST_ALICE2_EMU32_0,
+ 	EMU_DST_ALICE2_EMU32_1,
+--- linux-2.6.22.1.orig/sound/pci/emu10k1/emupcm.c
++++ linux-2.6.22.1/sound/pci/emu10k1/emupcm.c
+@@ -1233,24 +1233,26 @@
+ 	runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+ 	spin_lock_irq(&emu->reg_lock);
+ 	if (emu->card_capabilities->emu1010) {
+-		/* TODO 
++		/*  Nb. of channels has been increased to 16 */
++		/* TODO
+ 		 * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE
+ 		 * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ 		 * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ 		 * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000
+ 		 * rate_min = 44100,
+ 		 * rate_max = 192000,
+-		 * channels_min = 8,
+-		 * channels_max = 8,
++		 * channels_min = 16,
++		 * channels_max = 16,
+ 		 * Need to add mixer control to fix sample rate
+ 		 *                 
+-		 * There are 16 mono channels of 16bits each.
++		 * There are 32 mono channels of 16bits each.
+ 		 * 24bit Audio uses 2x channels over 16bit
+ 		 * 96kHz uses 2x channels over 48kHz
+ 		 * 192kHz uses 4x channels over 48kHz
+-		 * So, for 48kHz 24bit, one has 8 channels
+-		 * for 96kHz 24bit, one has 4 channels
+-		 * for 192kHz 24bit, one has 2 channels
++		 * So, for 48kHz 24bit, one has 16 channels
++		 * for 96kHz 24bit, one has 8 channels
++		 * for 192kHz 24bit, one has 4 channels
++		 *
+ 		 */
+ #if 1
+ 		switch (emu->emu1010.internal_clock) {
+@@ -1258,13 +1260,15 @@
+ 			/* For 44.1kHz */
+ 			runtime->hw.rates = SNDRV_PCM_RATE_44100;
+ 			runtime->hw.rate_min = runtime->hw.rate_max = 44100;
+-			runtime->hw.channels_min = runtime->hw.channels_max = 8;
++			runtime->hw.channels_min =
++				runtime->hw.channels_max = 16;
+ 			break;
+ 		case 1:
+ 			/* For 48kHz */
+ 			runtime->hw.rates = SNDRV_PCM_RATE_48000;
+ 			runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+-			runtime->hw.channels_min = runtime->hw.channels_max = 8;
++			runtime->hw.channels_min =
++				runtime->hw.channels_max = 16;
+ 			break;
+ 		};
+ #endif
+@@ -1282,7 +1286,7 @@
+ #endif
+ 		runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
+ 		/* efx_voices_mask[0] is expected to be zero
+- 		 * efx_voices_mask[1] is expected to have 16bits set
++ 		 * efx_voices_mask[1] is expected to have 32bits set
+ 		 */
+ 	} else {
+ 		runtime->hw.channels_min = runtime->hw.channels_max = 0;
+@@ -1787,11 +1791,24 @@
+ 	/* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */
+ 	if (emu->audigy) {
+ 		emu->efx_voices_mask[0] = 0;
+-		emu->efx_voices_mask[1] = 0xffff;
++		if (emu->card_capabilities->emu1010)
++			/* Pavel Hofman - 32 voices will be used for
++			 * capture (write mode) -
++			 * each bit = corresponding voice
++			 */
++			emu->efx_voices_mask[1] = 0xffffffff;
++		else
++			emu->efx_voices_mask[1] = 0xffff;
+ 	} else {
+ 		emu->efx_voices_mask[0] = 0xffff0000;
+ 		emu->efx_voices_mask[1] = 0;
+ 	}
++	/* For emu1010, the control has to set 32 upper bits (voices)
++	 * out of the 64 bits (voices) to true for the 16-channels capture
++	 * to work correctly. Correct A_FXWC2 initial value (0xffffffff)
++	 * is already defined but the snd_emu10k1_pcm_efx_voices_mask
++	 * control can override this register's value.
++	 */
+ 	kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu);
+ 	if (!kctl)
+ 		return -ENOMEM;
+--- linux-2.6.22.1.orig/sound/pci/ens1370.c
++++ linux-2.6.22.1/sound/pci/ens1370.c
+@@ -1607,8 +1607,8 @@
+ 	unsigned char rev;		/* revision */
+ };
+ 
+-static int __devinit es1371_quirk_lookup(struct ensoniq *ensoniq,
+-					 struct es1371_quirk *list)
++static int es1371_quirk_lookup(struct ensoniq *ensoniq,
++				struct es1371_quirk *list)
+ {
+ 	while (list->vid != (unsigned short)PCI_ANY_ID) {
+ 		if (ensoniq->pci->vendor == list->vid &&
+--- linux-2.6.22.1.orig/sound/pci/hda/hda_intel.c
++++ linux-2.6.22.1/sound/pci/hda/hda_intel.c
+@@ -341,6 +341,9 @@
+ 	unsigned int single_cmd :1;
+ 	unsigned int polling_mode :1;
+ 	unsigned int msi :1;
++
++	/* for debugging */
++	unsigned int last_cmd;	/* last issued command (to sync) */
+ };
+ 
+ /* driver types */
+@@ -466,18 +469,10 @@
+ }
+ 
+ /* send a command */
+-static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
+-			     unsigned int verb, unsigned int para)
++static int azx_corb_send_cmd(struct hda_codec *codec, u32 val)
+ {
+ 	struct azx *chip = codec->bus->private_data;
+ 	unsigned int wp;
+-	u32 val;
+-
+-	val = (u32)(codec->addr & 0x0f) << 28;
+-	val |= (u32)direct << 27;
+-	val |= (u32)nid << 20;
+-	val |= verb << 8;
+-	val |= para;
+ 
+ 	/* add command to corb */
+ 	wp = azx_readb(chip, CORBWP);
+@@ -538,12 +533,12 @@
+ 		}
+ 		if (! chip->rirb.cmds)
+ 			return chip->rirb.res; /* the last value */
+-		schedule_timeout_interruptible(1);
++		schedule_timeout(1);
+ 	} while (time_after_eq(timeout, jiffies));
+ 
+ 	if (chip->msi) {
+ 		snd_printk(KERN_WARNING "hda_intel: No response from codec, "
+-			   "disabling MSI...\n");
++			   "disabling MSI: last cmd=0x%08x\n", chip->last_cmd);
+ 		free_irq(chip->irq, chip);
+ 		chip->irq = -1;
+ 		pci_disable_msi(chip->pci);
+@@ -555,13 +550,15 @@
+ 
+ 	if (!chip->polling_mode) {
+ 		snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, "
+-			   "switching to polling mode...\n");
++			   "switching to polling mode: last cmd=0x%08x\n",
++			   chip->last_cmd);
+ 		chip->polling_mode = 1;
+ 		goto again;
+ 	}
+ 
+ 	snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
+-		   "switching to single_cmd mode...\n");
++		   "switching to single_cmd mode: last cmd=0x%08x\n",
++		   chip->last_cmd);
+ 	chip->rirb.rp = azx_readb(chip, RIRBWP);
+ 	chip->rirb.cmds = 0;
+ 	/* switch to single_cmd mode */
+@@ -581,20 +578,11 @@
+  */
+ 
+ /* send a command */
+-static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid,
+-			       int direct, unsigned int verb,
+-			       unsigned int para)
++static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
+ {
+ 	struct azx *chip = codec->bus->private_data;
+-	u32 val;
+ 	int timeout = 50;
+ 
+-	val = (u32)(codec->addr & 0x0f) << 28;
+-	val |= (u32)direct << 27;
+-	val |= (u32)nid << 20;
+-	val |= verb << 8;
+-	val |= para;
+-
+ 	while (timeout--) {
+ 		/* check ICB busy bit */
+ 		if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) {
+@@ -639,10 +627,19 @@
+ 			unsigned int para)
+ {
+ 	struct azx *chip = codec->bus->private_data;
++	u32 val;
++
++	val = (u32)(codec->addr & 0x0f) << 28;
++	val |= (u32)direct << 27;
++	val |= (u32)nid << 20;
++	val |= verb << 8;
++	val |= para;
++	chip->last_cmd = val;
++
+ 	if (chip->single_cmd)
+-		return azx_single_send_cmd(codec, nid, direct, verb, para);
++		return azx_single_send_cmd(codec, val);
+ 	else
+-		return azx_corb_send_cmd(codec, nid, direct, verb, para);
++		return azx_corb_send_cmd(codec, val);
+ }
+ 
+ /* get a response */
+@@ -1788,6 +1785,12 @@
+ 	{ 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */
+ 	{ 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */
+ 	{ 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */
++	{ 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */
++	{ 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */
++	{ 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
++	{ 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
++	{ 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
++	{ 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
+ 	{ 0, }
+ };
+ MODULE_DEVICE_TABLE(pci, azx_ids);
+--- linux-2.6.22.1.orig/sound/pci/hda/hda_proc.c
++++ linux-2.6.22.1/sound/pci/hda/hda_proc.c
+@@ -250,6 +250,12 @@
+ 	snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
+ 	snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
+ 	snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
++
++	if (codec->mfg)
++		snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg);
++	else
++		snd_iprintf(buffer, "No Modem Function Group found\n");
++
+ 	if (! codec->afg)
+ 		return;
+ 	snd_iprintf(buffer, "Default PCM:\n");
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_analog.c
++++ linux-2.6.22.1/sound/pci/hda/patch_analog.c
+@@ -1,7 +1,8 @@
+ /*
+- * HD audio interface patch for AD1981HD, AD1983, AD1986A, AD1988
++ * HD audio interface patch for AD1882, AD1884, AD1981HD, AD1983, AD1984,
++ *   AD1986A, AD1988
+  *
+- * Copyright (c) 2005 Takashi Iwai <tiwai at suse.de>
++ * Copyright (c) 2005-2007 Takashi Iwai <tiwai at suse.de>
+  *
+  *  This driver is free software; you can redistribute it and/or modify
+  *  it under the terms of the GNU General Public License as published by
+@@ -61,7 +62,7 @@
+ 	int num_channel_mode;
+ 
+ 	/* PCM information */
+-	struct hda_pcm pcm_rec[2];	/* used in alc_build_pcms() */
++	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
+ 
+ 	struct mutex amp_mutex;	/* PCM volume/mute control mutex */
+ 	unsigned int spdif_route;
+@@ -2775,11 +2776,634 @@
+ 
+ 
+ /*
++ * AD1884 / AD1984
++ *
++ * port-B - front line/mic-in
++ * port-E - aux in/out
++ * port-F - aux in/out
++ * port-C - rear line/mic-in
++ * port-D - rear line/hp-out
++ * port-A - front line/hp-out
++ *
++ * AD1984 = AD1884 + two digital mic-ins
++ *
++ * FIXME:
++ * For simplicity, we share the single DAC for both HP and line-outs
++ * right now.  The inidividual playbacks could be easily implemented,
++ * but no build-up framework is given, so far.
++ */
++
++static hda_nid_t ad1884_dac_nids[1] = {
++	0x04,
++};
++
++static hda_nid_t ad1884_adc_nids[2] = {
++	0x08, 0x09,
++};
++
++static hda_nid_t ad1884_capsrc_nids[2] = {
++	0x0c, 0x0d,
++};
++
++#define AD1884_SPDIF_OUT	0x02
++
++static struct hda_input_mux ad1884_capture_source = {
++	.num_items = 4,
++	.items = {
++		{ "Front Mic", 0x0 },
++		{ "Mic", 0x1 },
++		{ "CD", 0x2 },
++		{ "Mix", 0x3 },
++	},
++};
++
++static struct snd_kcontrol_new ad1884_base_mixers[] = {
++	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
++	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
++	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
++	/*
++	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
++	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
++	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
++	*/
++	HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* The multiple "Capture Source" controls confuse alsamixer
++		 * So call somewhat different..
++		 * FIXME: the controls appear in the "playback" view!
++		 */
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = ad198x_mux_enum_info,
++		.get = ad198x_mux_enum_get,
++		.put = ad198x_mux_enum_put,
++	},
++	/* SPDIF controls */
++	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
++		/* identical with ad1983 */
++		.info = ad1983_spdif_route_info,
++		.get = ad1983_spdif_route_get,
++		.put = ad1983_spdif_route_put,
++	},
++	{ } /* end */
++};
++
++static struct snd_kcontrol_new ad1984_dmic_mixers[] = {
++	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
++			     HDA_INPUT),
++	HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
++			   HDA_INPUT),
++	{ } /* end */
++};
++
++/*
++ * initialization verbs
++ */
++static struct hda_verb ad1884_init_verbs[] = {
++	/* DACs; mute as default */
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	/* Port-A (HP) mixer */
++	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Port-A pin */
++	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* HP selector - select DAC2 */
++	{0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
++	/* Port-D (Line-out) mixer */
++	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Port-D pin */
++	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Mono-out mixer */
++	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Mono-out pin */
++	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Mono selector */
++	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
++	/* Port-B (front mic) pin */
++	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Port-C (rear mic) pin */
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Analog mixer; mute as default */
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
++	/* Analog Mix output amp */
++	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
++	/* SPDIF output selector */
++	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
++	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
++	{ } /* end */
++};
++
++static int patch_ad1884(struct hda_codec *codec)
++{
++	struct ad198x_spec *spec;
++
++	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
++	if (spec == NULL)
++		return -ENOMEM;
++
++	mutex_init(&spec->amp_mutex);
++	codec->spec = spec;
++
++	spec->multiout.max_channels = 2;
++	spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
++	spec->multiout.dac_nids = ad1884_dac_nids;
++	spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
++	spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
++	spec->adc_nids = ad1884_adc_nids;
++	spec->capsrc_nids = ad1884_capsrc_nids;
++	spec->input_mux = &ad1884_capture_source;
++	spec->num_mixers = 1;
++	spec->mixers[0] = ad1884_base_mixers;
++	spec->num_init_verbs = 1;
++	spec->init_verbs[0] = ad1884_init_verbs;
++	spec->spdif_route = 0;
++
++	codec->patch_ops = ad198x_patch_ops;
++
++	return 0;
++}
++
++/*
++ * Lenovo Thinkpad T61/X61
++ */
++static struct hda_input_mux ad1984_thinkpad_capture_source = {
++	.num_items = 3,
++	.items = {
++		{ "Mic", 0x0 },
++		{ "Internal Mic", 0x1 },
++		{ "Mix", 0x3 },
++	},
++};
++
++static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
++	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
++	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
++	HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
++	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
++	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* The multiple "Capture Source" controls confuse alsamixer
++		 * So call somewhat different..
++		 * FIXME: the controls appear in the "playback" view!
++		 */
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = ad198x_mux_enum_info,
++		.get = ad198x_mux_enum_get,
++		.put = ad198x_mux_enum_put,
++	},
++	{ } /* end */
++};
++
++/* additional verbs */
++static struct hda_verb ad1984_thinkpad_init_verbs[] = {
++	/* Port-E (docking station mic) pin */
++	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* docking mic boost */
++	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Analog mixer - docking mic; mute as default */
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
++	/* enable EAPD bit */
++	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
++	{ } /* end */
++};
++
++/* Digial MIC ADC NID 0x05 + 0x06 */
++static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
++				   struct hda_codec *codec,
++				   unsigned int stream_tag,
++				   unsigned int format,
++				   struct snd_pcm_substream *substream)
++{
++	snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
++				   stream_tag, 0, format);
++	return 0;
++}
++
++static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
++				   struct hda_codec *codec,
++				   struct snd_pcm_substream *substream)
++{
++	snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
++				   0, 0, 0);
++	return 0;
++}
++
++static struct hda_pcm_stream ad1984_pcm_dmic_capture = {
++	.substreams = 2,
++	.channels_min = 2,
++	.channels_max = 2,
++	.nid = 0x05,
++	.ops = {
++		.prepare = ad1984_pcm_dmic_prepare,
++		.cleanup = ad1984_pcm_dmic_cleanup
++	},
++};
++
++static int ad1984_build_pcms(struct hda_codec *codec)
++{
++	struct ad198x_spec *spec = codec->spec;
++	struct hda_pcm *info;
++	int err;
++
++	err = ad198x_build_pcms(codec);
++	if (err < 0)
++		return err;
++
++	info = spec->pcm_rec + codec->num_pcms;
++	codec->num_pcms++;
++	info->name = "AD1984 Digital Mic";
++	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
++	return 0;
++}
++
++/* models */
++enum {
++	AD1984_BASIC,
++	AD1984_THINKPAD,
++	AD1984_MODELS
++};
++
++static const char *ad1984_models[AD1984_MODELS] = {
++	[AD1984_BASIC]		= "basic",
++	[AD1984_THINKPAD]	= "thinkpad",
++};
++
++static struct snd_pci_quirk ad1984_cfg_tbl[] = {
++	/* Lenovo Thinkpad T61/X61 */
++	SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD),
++	{}
++};
++
++static int patch_ad1984(struct hda_codec *codec)
++{
++	struct ad198x_spec *spec;
++	int board_config, err;
++
++	err = patch_ad1884(codec);
++	if (err < 0)
++		return err;
++	spec = codec->spec;
++	board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
++						  ad1984_models, ad1984_cfg_tbl);
++	switch (board_config) {
++	case AD1984_BASIC:
++		/* additional digital mics */
++		spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
++		codec->patch_ops.build_pcms = ad1984_build_pcms;
++		break;
++	case AD1984_THINKPAD:
++		spec->multiout.dig_out_nid = 0;
++		spec->input_mux = &ad1984_thinkpad_capture_source;
++		spec->mixers[0] = ad1984_thinkpad_mixers;
++		spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
++		break;
++	}
++	return 0;
++}
++
++
++/*
++ * AD1882
++ *
++ * port-A - front hp-out
++ * port-B - front mic-in
++ * port-C - rear line-in, shared surr-out (3stack)
++ * port-D - rear line-out
++ * port-E - rear mic-in, shared clfe-out (3stack)
++ * port-F - rear surr-out (6stack)
++ * port-G - rear clfe-out (6stack)
++ */
++
++static hda_nid_t ad1882_dac_nids[3] = {
++	0x04, 0x03, 0x05
++};
++
++static hda_nid_t ad1882_adc_nids[2] = {
++	0x08, 0x09,
++};
++
++static hda_nid_t ad1882_capsrc_nids[2] = {
++	0x0c, 0x0d,
++};
++
++#define AD1882_SPDIF_OUT	0x02
++
++/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
++static struct hda_input_mux ad1882_capture_source = {
++	.num_items = 5,
++	.items = {
++		{ "Front Mic", 0x1 },
++		{ "Mic", 0x4 },
++		{ "Line", 0x2 },
++		{ "CD", 0x3 },
++		{ "Mix", 0x7 },
++	},
++};
++
++static struct snd_kcontrol_new ad1882_base_mixers[] = {
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
++	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
++	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
++	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
++	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
++	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
++	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* The multiple "Capture Source" controls confuse alsamixer
++		 * So call somewhat different..
++		 * FIXME: the controls appear in the "playback" view!
++		 */
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = ad198x_mux_enum_info,
++		.get = ad198x_mux_enum_get,
++		.put = ad198x_mux_enum_put,
++	},
++	/* SPDIF controls */
++	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
++		/* identical with ad1983 */
++		.info = ad1983_spdif_route_info,
++		.get = ad1983_spdif_route_get,
++		.put = ad1983_spdif_route_put,
++	},
++	{ } /* end */
++};
++
++static struct snd_kcontrol_new ad1882_3stack_mixers[] = {
++	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		.name = "Channel Mode",
++		.info = ad198x_ch_mode_info,
++		.get = ad198x_ch_mode_get,
++		.put = ad198x_ch_mode_put,
++	},
++	{ } /* end */
++};
++
++static struct snd_kcontrol_new ad1882_6stack_mixers[] = {
++	HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
++	{ } /* end */
++};
++
++static struct hda_verb ad1882_ch2_init[] = {
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	{ } /* end */
++};
++
++static struct hda_verb ad1882_ch4_init[] = {
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	{ } /* end */
++};
++
++static struct hda_verb ad1882_ch6_init[] = {
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{ } /* end */
++};
++
++static struct hda_channel_mode ad1882_modes[3] = {
++	{ 2, ad1882_ch2_init },
++	{ 4, ad1882_ch4_init },
++	{ 6, ad1882_ch6_init },
++};
++
++/*
++ * initialization verbs
++ */
++static struct hda_verb ad1882_init_verbs[] = {
++	/* DACs; mute as default */
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	/* Port-A (HP) mixer */
++	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Port-A pin */
++	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* HP selector - select DAC2 */
++	{0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
++	/* Port-D (Line-out) mixer */
++	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Port-D pin */
++	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Mono-out mixer */
++	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	/* Mono-out pin */
++	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Port-B (front mic) pin */
++	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
++	/* Port-C (line-in) pin */
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
++	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
++	/* Port-C mixer - mute as input */
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	/* Port-E (mic-in) pin */
++	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
++	/* Port-E mixer - mute as input */
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	/* Port-F (surround) */
++	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Port-G (CLFE) */
++	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	/* Analog mixer; mute as default */
++	/* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
++	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
++	/* Analog Mix output amp */
++	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
++	/* SPDIF output selector */
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
++	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
++	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
++	{ } /* end */
++};
++
++/* models */
++enum {
++	AD1882_3STACK,
++	AD1882_6STACK,
++	AD1882_MODELS
++};
++
++static const char *ad1882_models[AD1986A_MODELS] = {
++	[AD1882_3STACK]		= "3stack",
++	[AD1882_6STACK]		= "6stack",
++};
++
++
++static int patch_ad1882(struct hda_codec *codec)
++{
++	struct ad198x_spec *spec;
++	int board_config;
++
++	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
++	if (spec == NULL)
++		return -ENOMEM;
++
++	mutex_init(&spec->amp_mutex);
++	codec->spec = spec;
++
++	spec->multiout.max_channels = 6;
++	spec->multiout.num_dacs = 3;
++	spec->multiout.dac_nids = ad1882_dac_nids;
++	spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
++	spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
++	spec->adc_nids = ad1882_adc_nids;
++	spec->capsrc_nids = ad1882_capsrc_nids;
++	spec->input_mux = &ad1882_capture_source;
++	spec->num_mixers = 1;
++	spec->mixers[0] = ad1882_base_mixers;
++	spec->num_init_verbs = 1;
++	spec->init_verbs[0] = ad1882_init_verbs;
++	spec->spdif_route = 0;
++
++	codec->patch_ops = ad198x_patch_ops;
++
++	/* override some parameters */
++	board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
++						  ad1882_models, NULL);
++	switch (board_config) {
++	default:
++	case AD1882_3STACK:
++		spec->num_mixers = 2;
++		spec->mixers[1] = ad1882_3stack_mixers;
++		spec->channel_mode = ad1882_modes;
++		spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
++		spec->need_dac_fix = 1;
++		spec->multiout.max_channels = 2;
++		spec->multiout.num_dacs = 1;
++		break;
++	case AD1882_6STACK:
++		spec->num_mixers = 2;
++		spec->mixers[1] = ad1882_6stack_mixers;
++		break;
++	}
++	return 0;
++}
++
++
++/*
+  * patch entries
+  */
+ struct hda_codec_preset snd_hda_preset_analog[] = {
++	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
++	{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
+ 	{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
+ 	{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
++	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+ 	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
+ 	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
+ 	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_atihdmi.c
++++ linux-2.6.22.1/sound/pci/hda/patch_atihdmi.c
+@@ -172,6 +172,7 @@
+  */
+ struct hda_codec_preset snd_hda_preset_atihdmi[] = {
+ 	{ .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
++	{ .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
+ 	{ .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
+ 	{ .id = 0x1002aa01, .name = "ATI R600 HDMI", .patch = patch_atihdmi },
+ 	{} /* terminator */
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_conexant.c
++++ linux-2.6.22.1/sound/pci/hda/patch_conexant.c
+@@ -801,7 +801,9 @@
+ 	SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP),
+ 	SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP),
+ 	SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP),
++	SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP),
+ 	SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU),
++	SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP),
+ 	SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP),
+ 	{}
+ };
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_realtek.c
++++ linux-2.6.22.1/sound/pci/hda/patch_realtek.c
+@@ -94,10 +94,18 @@
+ 	ALC262_HP_BPC_D7000_WF,
+ 	ALC262_BENQ_ED8,
+ 	ALC262_SONY_ASSAMD,
++	ALC262_BENQ_T31,
+ 	ALC262_AUTO,
+ 	ALC262_MODEL_LAST /* last tag */
+ };
+ 
++/* ALC268 models */
++enum {
++	ALC268_3ST,
++	ALC268_AUTO,
++	ALC268_MODEL_LAST /* last tag */
++};
++
+ /* ALC861 models */
+ enum {
+ 	ALC861_3ST,
+@@ -115,6 +123,7 @@
+ /* ALC861-VD models */
+ enum {
+ 	ALC660VD_3ST,
++	ALC660VD_3ST_DIG,
+ 	ALC861VD_3ST,
+ 	ALC861VD_3ST_DIG,
+ 	ALC861VD_6ST_DIG,
+@@ -144,6 +153,7 @@
+ 	ALC882_TARGA,
+ 	ALC882_ASUS_A7J,
+ 	ALC885_MACPRO,
++	ALC885_IMAC24,
+ 	ALC882_AUTO,
+ 	ALC882_MODEL_LAST,
+ };
+@@ -163,6 +173,8 @@
+ 	ALC883_LENOVO_101E_2ch,
+ 	ALC883_LENOVO_NB0763,
+ 	ALC888_LENOVO_MS7195_DIG,		
++	ALC888_6ST_HP,
++	ALC888_3ST_HP,
+ 	ALC883_AUTO,
+ 	ALC883_MODEL_LAST,
+ };
+@@ -713,6 +725,38 @@
+ }
+ 
+ /*
++ * Fix-up pin default configurations
++ */
++
++struct alc_pincfg {
++	hda_nid_t nid;
++	u32 val;
++};
++
++static void alc_fix_pincfg(struct hda_codec *codec,
++			   const struct snd_pci_quirk *quirk,
++			   const struct alc_pincfg **pinfix)
++{
++	const struct alc_pincfg *cfg;
++
++	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
++	if (!quirk)
++		return;
++
++	cfg = pinfix[quirk->value];
++	for (; cfg->nid; cfg++) {
++		int i;
++		u32 val = cfg->val;
++		for (i = 0; i < 4; i++) {
++			snd_hda_codec_write(codec, cfg->nid, 0,
++				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
++				    val & 0xff);
++			val >>= 8;
++		}
++	}
++}
++
++/*
+  * ALC880 3-stack model
+  *
+  * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
+@@ -1878,31 +1922,53 @@
+  * Pin assignment:
+  *   Speaker-out: 0x14
+  *   Mic-In: 0x18
+- *   Built-in Mic-In: 0x19 (?)
+- *   HP-Out: 0x1b
++ *   Built-in Mic-In: 0x19
++ *   Line-In: 0x1b
++ *   HP-Out: 0x1a
+  *   SPDIF-Out: 0x1e
+  */
+ 
+-/* seems analog CD is not working */
+ static struct hda_input_mux alc880_lg_lw_capture_source = {
+-	.num_items = 2,
++	.num_items = 3,
+ 	.items = {
+ 		{ "Mic", 0x0 },
+ 		{ "Internal Mic", 0x1 },
++		{ "Line In", 0x2 },
+ 	},
+ };
+ 
++#define alc880_lg_lw_modes alc880_threestack_modes
++
+ static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
+-	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+-	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
++	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
++	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ 	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		.name = "Channel Mode",
++		.info = alc_ch_mode_info,
++		.get = alc_ch_mode_get,
++		.put = alc_ch_mode_put,
++	},
+ 	{ } /* end */
+ };
+ 
+ static struct hda_verb alc880_lg_lw_init_verbs[] = {
++	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
++	{0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
++	{0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
++
+ 	/* set capture source to mic-in */
+ 	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+@@ -1912,7 +1978,6 @@
+ 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ 	/* HP-out */
+-	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+ 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ 	/* mic-in to input */
+@@ -2856,11 +2921,11 @@
+ 		.mixers = { alc880_lg_lw_mixer },
+ 		.init_verbs = { alc880_volume_init_verbs,
+ 				alc880_lg_lw_init_verbs },
+-		.num_dacs = 1,
++		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ 		.dac_nids = alc880_dac_nids,
+ 		.dig_out_nid = ALC880_DIGOUT_NID,
+-		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+-		.channel_mode = alc880_2_jack_modes,
++		.num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
++		.channel_mode = alc880_lg_lw_modes,
+ 		.input_mux = &alc880_lg_lw_capture_source,
+ 		.unsol_event = alc880_lg_lw_unsol_event,
+ 		.init_hook = alc880_lg_lw_automute,
+@@ -5054,6 +5119,60 @@
+ 	{ }
+ };
+ 
++/* iMac 24 mixer. */
++static struct snd_kcontrol_new alc885_imac24_mixer[] = {
++	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
++	{ } /* end */
++};
++
++/* iMac 24 init verbs. */
++static struct hda_verb alc885_imac24_init_verbs[] = {
++	/* Internal speakers: output 0 (0x0c) */
++	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
++	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
++	/* Internal speakers: output 0 (0x0c) */
++	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
++	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
++	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
++	/* Headphone: output 0 (0x0c) */
++	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
++	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
++	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
++	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
++	/* Front Mic: input vref at 80% */
++	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
++	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{ }
++};
++
++/* Toggle speaker-output according to the hp-jack state */
++static void alc885_imac24_automute(struct hda_codec *codec)
++{
++ 	unsigned int present;
++
++ 	present = snd_hda_codec_read(codec, 0x14, 0,
++				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
++	snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0,
++				 0x80, present ? 0x80 : 0);
++	snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0,
++				 0x80, present ? 0x80 : 0);
++	snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
++				 0x80, present ? 0x80 : 0);
++	snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
++				 0x80, present ? 0x80 : 0);
++}
++
++/* Processes unsolicited events. */
++static void alc885_imac24_unsol_event(struct hda_codec *codec,
++				      unsigned int res)
++{
++	/* Headphone insertion or removal. */
++	if ((res >> 26) == ALC880_HP_EVENT)
++		alc885_imac24_automute(codec);
++}
++
+ static struct hda_verb alc882_targa_verbs[] = {
+ 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+@@ -5274,6 +5393,7 @@
+ 	[ALC882_ARIMA]		= "arima",
+ 	[ALC882_W2JC]		= "w2jc",
+ 	[ALC885_MACPRO]		= "macpro",
++	[ALC885_IMAC24]		= "imac24",
+ 	[ALC882_AUTO]		= "auto",
+ };
+ 
+@@ -5284,6 +5404,7 @@
+ 	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
+ 	SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
+ 	SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
++	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+ 	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
+ 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+ 	{}
+@@ -5345,6 +5466,19 @@
+ 		.channel_mode = alc882_ch_modes,
+ 		.input_mux = &alc882_capture_source,
+ 	},
++	[ALC885_IMAC24] = {
++		.mixers = { alc885_imac24_mixer },
++		.init_verbs = { alc885_imac24_init_verbs },
++		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
++		.dac_nids = alc882_dac_nids,
++		.dig_out_nid = ALC882_DIGOUT_NID,
++		.dig_in_nid = ALC882_DIGIN_NID,
++		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
++		.channel_mode = alc882_ch_modes,
++		.input_mux = &alc882_capture_source,
++		.unsol_event = alc885_imac24_unsol_event,
++		.init_hook = alc885_imac24_automute,
++	},
+ 	[ALC882_TARGA] = {
+ 		.mixers = { alc882_targa_mixer, alc882_chmode_mixer,
+ 			    alc882_capture_mixer },
+@@ -5379,6 +5513,29 @@
+ 
+ 
+ /*
++ * Pin config fixes
++ */
++enum { 
++	PINFIX_ABIT_AW9D_MAX
++};
++
++static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
++	{ 0x15, 0x01080104 }, /* side */
++	{ 0x16, 0x01011012 }, /* rear */
++	{ 0x17, 0x01016011 }, /* clfe */
++	{ }
++};
++
++static const struct alc_pincfg *alc882_pin_fixes[] = {
++	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
++};
++
++static struct snd_pci_quirk alc882_pinfix_tbl[] = {
++	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
++	{}
++};
++
++/*
+  * BIOS auto configuration
+  */
+ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
+@@ -5494,6 +5651,9 @@
+ 		case 0x106b0c00: /* Mac Pro */
+ 			board_config = ALC885_MACPRO;
+ 			break;
++		case 0x106b1000: /* iMac 24 */
++			board_config = ALC885_IMAC24;
++			break;
+ 		default:
+ 			printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
+ 		       			 "trying auto-probe from BIOS...\n");
+@@ -5501,6 +5661,8 @@
+ 		}
+ 	}
+ 
++	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
++
+ 	if (board_config == ALC882_AUTO) {
+ 		/* automatic parse from the BIOS config */
+ 		err = alc882_parse_auto_config(codec);
+@@ -5518,7 +5680,7 @@
+ 	if (board_config != ALC882_AUTO)
+ 		setup_preset(spec, &alc882_presets[board_config]);
+ 
+-	if (board_config == ALC885_MACPRO) {
++	if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) {
+ 		alc882_gpio_mute(codec, 0, 0);
+ 		alc882_gpio_mute(codec, 1, 0);
+ 	}
+@@ -5995,6 +6157,84 @@
+ 	{ } /* end */
+ };	
+ 
++static struct snd_kcontrol_new alc888_6st_hp_mixer[] = {
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
++	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
++	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
++	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
++	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
++	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
++	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
++	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
++	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = alc883_mux_enum_info,
++		.get = alc883_mux_enum_get,
++		.put = alc883_mux_enum_put,
++	},
++	{ } /* end */
++};
++
++static struct snd_kcontrol_new alc888_3st_hp_mixer[] = {
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
++	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
++	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
++	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
++	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
++	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
++	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
++	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
++	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
++	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
++	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
++	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = alc883_mux_enum_info,
++		.get = alc883_mux_enum_get,
++		.put = alc883_mux_enum_put,
++	},
++	{ } /* end */
++};
++
+ static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+ 	{
+ 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+@@ -6126,6 +6366,42 @@
+ 	{ } /* end */
+ };
+ 
++static struct hda_verb alc888_6st_hp_verbs[] = {
++	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Front: output 0 (0x0c) */
++	{0x15, AC_VERB_SET_CONNECT_SEL, 0x02},	/* Rear : output 2 (0x0e) */
++	{0x16, AC_VERB_SET_CONNECT_SEL, 0x01},	/* CLFE : output 1 (0x0d) */
++	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},	/* Side : output 3 (0x0f) */
++	{ }
++};
++
++static struct hda_verb alc888_3st_hp_verbs[] = {
++	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Front: output 0 (0x0c) */
++	{0x18, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Rear : output 1 (0x0d) */
++	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},	/* CLFE : output 2 (0x0e) */
++	{ }
++};
++
++static struct hda_verb alc888_3st_hp_2ch_init[] = {
++	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
++	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
++	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
++	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
++	{ }
++};
++
++static struct hda_verb alc888_3st_hp_6ch_init[] = {
++	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
++	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
++	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
++	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
++	{ }
++};
++
++static struct hda_channel_mode alc888_3st_hp_modes[2] = {
++	{ 2, alc888_3st_hp_2ch_init },
++	{ 6, alc888_3st_hp_6ch_init },
++};
++
+ /* toggle front-jack and RCA according to the hp-jack state */
+ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
+ {
+@@ -6368,11 +6644,14 @@
+ 	[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
+ 	[ALC883_LENOVO_NB0763]	= "lenovo-nb0763",
+ 	[ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
++	[ALC888_6ST_HP]		= "6stack-hp",
++	[ALC888_3ST_HP]		= "3stack-hp",
+ 	[ALC883_AUTO]		= "auto",
+ };
+ 
+ static struct snd_pci_quirk alc883_cfg_tbl[] = {
+ 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
++	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
+ 	SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+ 	SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
+ 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+@@ -6381,6 +6660,8 @@
+ 	SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
+ 	SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+ 	SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
++	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
++	SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
+ 	SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+ 	SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
+ 	SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
+@@ -6400,6 +6681,9 @@
+ 	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
+ 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ 	SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
++	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP),
++	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
++	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
+ 	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+ 	{}
+ };
+@@ -6584,6 +6868,31 @@
+ 		.unsol_event = alc883_lenovo_ms7195_unsol_event,
+ 		.init_hook = alc888_lenovo_ms7195_front_automute,
+ 	},	
++	[ALC888_6ST_HP] = {
++		.mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer },
++		.init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs },
++		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
++		.dac_nids = alc883_dac_nids,
++		.dig_out_nid = ALC883_DIGOUT_NID,
++		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
++		.adc_nids = alc883_adc_nids,
++		.dig_in_nid = ALC883_DIGIN_NID,
++		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
++		.channel_mode = alc883_sixstack_modes,
++		.input_mux = &alc883_capture_source,
++	},
++	[ALC888_3ST_HP] = {
++		.mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer },
++		.init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
++		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
++		.dac_nids = alc883_dac_nids,
++		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
++		.adc_nids = alc883_adc_nids,
++		.num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
++		.channel_mode = alc888_3st_hp_modes,
++		.need_dac_fix = 1,
++		.input_mux = &alc883_capture_source,
++	},
+ };
+ 
+ 
+@@ -6857,7 +7166,16 @@
+ 	{ } /* end */
+ };
+ 
+-
++static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
++	HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
++	HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
++	{ } /* end */
++};
+ 
+ #define alc262_capture_mixer		alc882_capture_mixer
+ #define alc262_capture_alt_mixer	alc882_capture_alt_mixer
+@@ -7189,6 +7507,15 @@
+ 	{}
+ };
+ 
++static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
++	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
++	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
++
++	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
++	{0x20, AC_VERB_SET_PROC_COEF,  0x3050},
++	{}
++};
++
+ /* add playback controls from the parsed DAC table */
+ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
+ 					     const struct auto_pin_cfg *cfg)
+@@ -7584,7 +7911,8 @@
+ 	[ALC262_HP_BPC]		= "hp-bpc",
+ 	[ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
+ 	[ALC262_BENQ_ED8]	= "benq",
+-	[ALC262_BENQ_ED8]	= "sony-assamd",
++	[ALC262_BENQ_T31]	= "benq-t31",
++	[ALC262_SONY_ASSAMD]	= "sony-assamd",
+ 	[ALC262_AUTO]		= "auto",
+ };
+ 
+@@ -7592,8 +7920,12 @@
+ 	SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
+ 	SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
+ 	SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
++	SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC),
++	SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC),
+ 	SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
++	SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC),
+ 	SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
++	SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC),
+ 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ 	SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
+@@ -7606,6 +7938,7 @@
+ 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+ 	SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
+ 	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
++	SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+ 	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
+ 	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ 	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+@@ -7710,6 +8043,17 @@
+ 		.channel_mode = alc262_modes,
+ 		.input_mux = &alc262_capture_source,
+ 		.unsol_event = alc262_hippo_unsol_event,
++	},
++	[ALC262_BENQ_T31] = {
++		.mixers = { alc262_benq_t31_mixer },
++		.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs },
++		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
++		.dac_nids = alc262_dac_nids,
++		.hp_nid = 0x03,
++		.num_channel_mode = ARRAY_SIZE(alc262_modes),
++		.channel_mode = alc262_modes,
++		.input_mux = &alc262_capture_source,
++		.unsol_event = alc262_hippo_unsol_event,
+ 	},	
+ };
+ 
+@@ -7800,31 +8144,540 @@
+ }
+ 
+ /*
+- *  ALC861 channel source setting (2/6 channel selection for 3-stack)
++ *  ALC268 channel source setting (2 channel)
+  */
++#define ALC268_DIGOUT_NID	ALC880_DIGOUT_NID
++#define alc268_modes		alc260_modes
++	
++static hda_nid_t alc268_dac_nids[2] = {
++	/* front, hp */
++	0x02, 0x03
++};
+ 
+-/*
+- * set the path ways for 2 channel output
+- * need to set the codec line out and mic 1 pin widgets to inputs
+- */
+-static struct hda_verb alc861_threestack_ch2_init[] = {
+-	/* set pin widget 1Ah (line in) for input */
+-	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+-	/* set pin widget 18h (mic1/2) for input, for mic also enable
+-	 * the vref
+-	 */
+-	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
++static hda_nid_t alc268_adc_nids[2] = {
++	/* ADC0-1 */
++	0x08, 0x07
++};
+ 
+-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+-#if 0
+-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+-#endif
+-	{ } /* end */
++static hda_nid_t alc268_adc_nids_alt[1] = {
++	/* ADC0 */
++	0x08
+ };
+-/*
+- * 6ch mode
+- * need to set the codec line out and mic 1 pin widgets to outputs
++
++static struct snd_kcontrol_new alc268_base_mixer[] = {
++	/* output mixer control */
++	HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
++	{ }
++};
++
++/*
++ * generic initialization of ADC, input mixers and output mixers
++ */
++static struct hda_verb alc268_base_init_verbs[] = {
++	/* Unmute DAC0-1 and set vol = 0 */
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++
++	/*
++	 * Set up output mixers (0x0c - 0x0e)
++	 */
++	/* set vol=0 to output mixers */
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++        {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
++
++	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++
++	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
++	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
++	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
++	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
++	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
++	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++
++	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
++
++	/* FIXME: use matrix-type input source selection */
++	/* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */
++	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
++	/* Input mixer2 */
++	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
++	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
++	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
++	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
++
++	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
++	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
++	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
++	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
++	{ }
++};
++
++/*
++ * generic initialization of ADC, input mixers and output mixers
++ */
++static struct hda_verb alc268_volume_init_verbs[] = {
++	/* set output DAC */
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++
++	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
++	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
++	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
++
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
++	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
++
++	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
++
++	/* set PCBEEP vol = 0 */
++	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))},
++
++	{ }
++};
++
++#define alc268_mux_enum_info alc_mux_enum_info
++#define alc268_mux_enum_get alc_mux_enum_get
++
++static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol,
++			       struct snd_ctl_elem_value *ucontrol)
++{
++	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
++	struct alc_spec *spec = codec->spec;
++	const struct hda_input_mux *imux = spec->input_mux;
++	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
++	static hda_nid_t capture_mixers[3] = { 0x23, 0x24 };
++	hda_nid_t nid = capture_mixers[adc_idx];
++	unsigned int *cur_val = &spec->cur_mux[adc_idx];
++	unsigned int i, idx;
++
++	idx = ucontrol->value.enumerated.item[0];
++	if (idx >= imux->num_items)
++		idx = imux->num_items - 1;
++	if (*cur_val == idx && !codec->in_resume)
++		return 0;
++	for (i = 0; i < imux->num_items; i++) {
++		unsigned int v = (i == idx) ? 0x7000 : 0x7080;
++		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
++				    v | (imux->items[i].index << 8));
++                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
++				    idx );
++	}
++	*cur_val = idx;
++	return 1;
++}
++
++static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
++	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* The multiple "Capture Source" controls confuse alsamixer
++		 * So call somewhat different..
++		 * FIXME: the controls appear in the "playback" view!
++		 */
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 1,
++		.info = alc268_mux_enum_info,
++		.get = alc268_mux_enum_get,
++		.put = alc268_mux_enum_put,
++	},
++	{ } /* end */
++};
++
++static struct snd_kcontrol_new alc268_capture_mixer[] = {
++	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
++	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
++	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
++	{
++		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++		/* The multiple "Capture Source" controls confuse alsamixer
++		 * So call somewhat different..
++		 * FIXME: the controls appear in the "playback" view!
++		 */
++		/* .name = "Capture Source", */
++		.name = "Input Source",
++		.count = 2,
++		.info = alc268_mux_enum_info,
++		.get = alc268_mux_enum_get,
++		.put = alc268_mux_enum_put,
++	},
++	{ } /* end */
++};
++
++static struct hda_input_mux alc268_capture_source = {
++	.num_items = 4,
++	.items = {
++		{ "Mic", 0x0 },
++		{ "Front Mic", 0x1 },
++		{ "Line", 0x2 },
++		{ "CD", 0x3 },
++	},
++};
++
++/* create input playback/capture controls for the given pin */
++static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
++				    const char *ctlname, int idx)
++{
++	char name[32];
++	int err;
++
++	sprintf(name, "%s Playback Volume", ctlname);
++	if (nid == 0x14) {
++		err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
++				  HDA_COMPOSE_AMP_VAL(0x02, 3, idx,
++						      HDA_OUTPUT));
++		if (err < 0)
++			return err;
++	} else if (nid == 0x15) {
++		err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
++				  HDA_COMPOSE_AMP_VAL(0x03, 3, idx,
++						      HDA_OUTPUT));
++		if (err < 0)
++			return err;
++	} else
++		return -1;
++	sprintf(name, "%s Playback Switch", ctlname);
++	err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
++			  HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
++	if (err < 0)
++		return err;
++	return 0;
++}
++
++/* add playback controls from the parsed DAC table */
++static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
++					     const struct auto_pin_cfg *cfg)
++{
++	hda_nid_t nid;
++	int err;
++
++	spec->multiout.num_dacs = 2;	/* only use one dac */
++	spec->multiout.dac_nids = spec->private_dac_nids;
++	spec->multiout.dac_nids[0] = 2;
++	spec->multiout.dac_nids[1] = 3;
++
++	nid = cfg->line_out_pins[0];
++	if (nid)
++		alc268_new_analog_output(spec, nid, "Front", 0);	
++
++	nid = cfg->speaker_pins[0];
++	if (nid == 0x1d) {
++		err = add_control(spec, ALC_CTL_WIDGET_VOL,
++				  "Speaker Playback Volume",
++				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
++		if (err < 0)
++			return err;
++	}
++	nid = cfg->hp_pins[0];
++	if (nid)
++		alc268_new_analog_output(spec, nid, "Headphone", 0);
++
++	nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
++	if (nid == 0x16) {
++		err = add_control(spec, ALC_CTL_WIDGET_MUTE,
++				  "Mono Playback Switch",
++				  HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT));
++		if (err < 0)
++			return err;
++	}
++	return 0;	
++}
++
++/* create playback/capture controls for input pins */
++static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
++						const struct auto_pin_cfg *cfg)
++{
++	struct hda_input_mux *imux = &spec->private_imux;
++	int i, idx1;
++
++	for (i = 0; i < AUTO_PIN_LAST; i++) {
++		switch(cfg->input_pins[i]) {
++		case 0x18:
++			idx1 = 0;	/* Mic 1 */
++			break;
++		case 0x19:
++			idx1 = 1;	/* Mic 2 */
++			break;
++		case 0x1a:
++			idx1 = 2;	/* Line In */
++			break;
++		case 0x1c:	
++			idx1 = 3;	/* CD */
++			break;
++		default:
++			continue;
++		}
++		imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
++		imux->items[imux->num_items].index = idx1;
++		imux->num_items++;	
++	}
++	return 0;
++}
++
++static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
++{
++	struct alc_spec *spec = codec->spec;
++	hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
++	hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
++	hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
++	unsigned int	dac_vol1, dac_vol2;
++
++	if (speaker_nid) {
++		snd_hda_codec_write(codec, speaker_nid, 0,
++				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
++		snd_hda_codec_write(codec, 0x0f, 0,
++				    AC_VERB_SET_AMP_GAIN_MUTE,
++				    AMP_IN_UNMUTE(1));
++		snd_hda_codec_write(codec, 0x10, 0,
++				    AC_VERB_SET_AMP_GAIN_MUTE,
++				    AMP_IN_UNMUTE(1));
++	} else {
++		snd_hda_codec_write(codec, 0x0f, 0,
++				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
++		snd_hda_codec_write(codec, 0x10, 0,
++				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
++	}
++
++	dac_vol1 = dac_vol2 = 0xb000 | 0x40;	/* set max volume  */
++	if (line_nid == 0x14)	
++		dac_vol2 = AMP_OUT_ZERO;
++	else if (line_nid == 0x15)
++		dac_vol1 = AMP_OUT_ZERO;
++	if (hp_nid == 0x14)	
++		dac_vol2 = AMP_OUT_ZERO;
++	else if (hp_nid == 0x15)
++		dac_vol1 = AMP_OUT_ZERO;
++	if (line_nid != 0x16 || hp_nid != 0x16 ||
++	    spec->autocfg.line_out_pins[1] != 0x16 ||
++	    spec->autocfg.line_out_pins[2] != 0x16)
++		dac_vol1 = dac_vol2 = AMP_OUT_ZERO;
++
++	snd_hda_codec_write(codec, 0x02, 0,
++			    AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1);
++	snd_hda_codec_write(codec, 0x03, 0,
++			    AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
++}
++
++/* pcm configuration: identiacal with ALC880 */
++#define alc268_pcm_analog_playback	alc880_pcm_analog_playback
++#define alc268_pcm_analog_capture	alc880_pcm_analog_capture
++#define alc268_pcm_digital_playback	alc880_pcm_digital_playback
++
++/*
++ * BIOS auto configuration
++ */
++static int alc268_parse_auto_config(struct hda_codec *codec)
++{
++	struct alc_spec *spec = codec->spec;
++	int err;
++	static hda_nid_t alc268_ignore[] = { 0 };
++
++	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
++					   alc268_ignore);
++	if (err < 0)
++		return err;
++	if (!spec->autocfg.line_outs)
++		return 0; /* can't find valid BIOS pin config */
++
++	err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
++	if (err < 0)
++		return err;
++	err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg);
++	if (err < 0)
++		return err;
++
++	spec->multiout.max_channels = 2;
++
++	/* digital only support output */
++	if (spec->autocfg.dig_out_pin)
++		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
++
++	if (spec->kctl_alloc)
++		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
++
++	spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs;
++	spec->num_mux_defs = 1;
++	spec->input_mux = &spec->private_imux;
++
++	return 1;
++}
++
++#define alc268_auto_init_multi_out	alc882_auto_init_multi_out
++#define alc268_auto_init_hp_out		alc882_auto_init_hp_out
++#define alc268_auto_init_analog_input	alc882_auto_init_analog_input
++
++/* init callback for auto-configuration model -- overriding the default init */
++static void alc268_auto_init(struct hda_codec *codec)
++{
++	alc268_auto_init_multi_out(codec);
++	alc268_auto_init_hp_out(codec);
++	alc268_auto_init_mono_speaker_out(codec);
++	alc268_auto_init_analog_input(codec);
++}
++
++/*
++ * configuration and preset
++ */
++static const char *alc268_models[ALC268_MODEL_LAST] = {
++	[ALC268_3ST]		= "3stack",
++	[ALC268_AUTO]		= "auto",
++};
++
++static struct snd_pci_quirk alc268_cfg_tbl[] = {
++	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
++	{}
++};
++
++static struct alc_config_preset alc268_presets[] = {
++	[ALC268_3ST] = {
++		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
++		.init_verbs = { alc268_base_init_verbs },
++		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
++		.dac_nids = alc268_dac_nids,
++                .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
++                .adc_nids = alc268_adc_nids_alt,
++		.hp_nid = 0x03,
++		.dig_out_nid = ALC268_DIGOUT_NID,
++		.num_channel_mode = ARRAY_SIZE(alc268_modes),
++		.channel_mode = alc268_modes,
++		.input_mux = &alc268_capture_source,
++	},
++};
++
++static int patch_alc268(struct hda_codec *codec)
++{
++	struct alc_spec *spec;
++	int board_config;
++	int err;
++
++	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
++	if (spec == NULL)
++		return -ENOMEM;
++
++	codec->spec = spec;
++
++	board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST,
++						  alc268_models,
++						  alc268_cfg_tbl);
++
++	if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
++		printk(KERN_INFO "hda_codec: Unknown model for ALC268, "
++		       "trying auto-probe from BIOS...\n");
++		board_config = ALC268_AUTO;
++	}
++
++	if (board_config == ALC268_AUTO) {
++		/* automatic parse from the BIOS config */
++		err = alc268_parse_auto_config(codec);
++		if (err < 0) {
++			alc_free(codec);
++			return err;
++		} else if (!err) {
++			printk(KERN_INFO
++			       "hda_codec: Cannot set up configuration "
++			       "from BIOS.  Using base mode...\n");
++			board_config = ALC268_3ST;
++		}
++	}
++
++	if (board_config != ALC268_AUTO)
++		setup_preset(spec, &alc268_presets[board_config]);
++
++	spec->stream_name_analog = "ALC268 Analog";
++	spec->stream_analog_playback = &alc268_pcm_analog_playback;
++	spec->stream_analog_capture = &alc268_pcm_analog_capture;
++
++	spec->stream_name_digital = "ALC268 Digital";
++	spec->stream_digital_playback = &alc268_pcm_digital_playback;
++
++	if (board_config == ALC268_AUTO) {
++		if (!spec->adc_nids && spec->input_mux) {
++			/* check whether NID 0x07 is valid */
++			unsigned int wcap = get_wcaps(codec, 0x07);
++
++			/* get type */
++			wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
++			if (wcap != AC_WID_AUD_IN) {
++				spec->adc_nids = alc268_adc_nids_alt;
++				spec->num_adc_nids =
++					ARRAY_SIZE(alc268_adc_nids_alt);
++				spec->mixers[spec->num_mixers] =
++					alc268_capture_alt_mixer;
++				spec->num_mixers++;
++			} else {
++				spec->adc_nids = alc268_adc_nids;
++				spec->num_adc_nids =
++					ARRAY_SIZE(alc268_adc_nids);
++				spec->mixers[spec->num_mixers] =
++					alc268_capture_mixer;
++				spec->num_mixers++;
++			}
++		}
++	}
++	codec->patch_ops = alc_patch_ops;
++	if (board_config == ALC268_AUTO)
++		spec->init_hook = alc268_auto_init;
++		
++	return 0;
++}
++
++/*
++ *  ALC861 channel source setting (2/6 channel selection for 3-stack)
++ */
++
++/*
++ * set the path ways for 2 channel output
++ * need to set the codec line out and mic 1 pin widgets to inputs
++ */
++static struct hda_verb alc861_threestack_ch2_init[] = {
++	/* set pin widget 1Ah (line in) for input */
++	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
++	/* set pin widget 18h (mic1/2) for input, for mic also enable
++	 * the vref
++	 */
++	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
++
++	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
++#if 0
++	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
++	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
++#endif
++	{ } /* end */
++};
++/*
++ * 6ch mode
++ * need to set the codec line out and mic 1 pin widgets to outputs
+  */
+ static struct hda_verb alc861_threestack_ch6_init[] = {
+ 	/* set pin widget 1Ah (line in) for output (Back Surround)*/
+@@ -8767,13 +9620,21 @@
+ 	SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ 	SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ 	SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
++	SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
+ 	SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
++	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
+ 	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
+-	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA),
++	/* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
++	 *        Any other models that need this preset?
++	 */
++	/* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
+ 	SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
++	SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31),
+ 	SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
+ 	SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
+ 	SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
++	SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
++	SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
+ 	{}
+ };
+ 
+@@ -9464,6 +10325,7 @@
+  */
+ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
+ 	[ALC660VD_3ST]		= "3stack-660",
++	[ALC660VD_3ST_DIG]= "3stack-660-digout",
+ 	[ALC861VD_3ST]		= "3stack",
+ 	[ALC861VD_3ST_DIG]	= "3stack-digout",
+ 	[ALC861VD_6ST_DIG]	= "6stack-digout",
+@@ -9475,7 +10337,7 @@
+ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
+ 	SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
+ 	SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+-	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST),
++	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
+ 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
+ 	SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
+ 
+@@ -9483,6 +10345,7 @@
+ 	SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
+ 	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
+ 	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
++	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+ 	{}
+ };
+ 
+@@ -9499,6 +10362,19 @@
+ 		.channel_mode = alc861vd_3stack_2ch_modes,
+ 		.input_mux = &alc861vd_capture_source,
+ 	},
++	[ALC660VD_3ST_DIG] = {
++		.mixers = { alc861vd_3st_mixer },
++		.init_verbs = { alc861vd_volume_init_verbs,
++				 alc861vd_3stack_init_verbs },
++		.num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
++		.dac_nids = alc660vd_dac_nids,
++		.dig_out_nid = ALC861VD_DIGOUT_NID,
++		.num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
++		.adc_nids = alc861vd_adc_nids,
++		.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
++		.channel_mode = alc861vd_3stack_2ch_modes,
++		.input_mux = &alc861vd_capture_source,
++	},
+ 	[ALC861VD_3ST] = {
+ 		.mixers = { alc861vd_3st_mixer },
+ 		.init_verbs = { alc861vd_volume_init_verbs,
+@@ -10420,7 +11296,7 @@
+ 	for (i = 0; i < cfg->line_outs; i++) {
+ 		if (!spec->multiout.dac_nids[i])
+ 			continue;
+-		nid = alc880_idx_to_dac(i);
++		nid = alc880_idx_to_mixer(i);
+ 		if (i == 2) {
+ 			/* Center/LFE */
+ 			err = add_control(spec, ALC_CTL_WIDGET_VOL,
+@@ -10643,14 +11519,10 @@
+ 	spec->num_mux_defs = 1;
+ 	spec->input_mux = &spec->private_imux;
+ 	
+-	if (err < 0)
+-		return err;
+-	else if (err > 0)
+-		/* hack - override the init verbs */
+-		spec->init_verbs[0] = alc662_auto_init_verbs;
++	spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
+ 	spec->mixers[spec->num_mixers] = alc662_capture_mixer;
+ 	spec->num_mixers++;
+-	return err;
++	return 1;
+ }
+ 
+ /* additional initialization for auto-configuration model */
+@@ -10687,7 +11559,7 @@
+ 		if (err < 0) {
+ 			alc_free(codec);
+ 			return err;
+-		} else if (err) {
++		} else if (!err) {
+ 			printk(KERN_INFO
+ 			       "hda_codec: Cannot set up configuration "
+ 			       "from BIOS.  Using base mode...\n");
+@@ -10724,6 +11596,7 @@
+ struct hda_codec_preset snd_hda_preset_realtek[] = {
+ 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
+ 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
++	{ .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
+ 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
+ 	  .patch = patch_alc861 },
+ 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_si3054.c
++++ linux-2.6.22.1/sound/pci/hda/patch_si3054.c
+@@ -304,8 +304,12 @@
+  	{ .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 },
+  	{ .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
+  	{ .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 },
++	/* VIA HDA on Clevo m540 */
++	{ .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 },
+ 	/* Asus A8J Modem (SM56) */
+ 	{ .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 },
++	/* LG LW20 modem */
++	{ .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 },
+ 	{}
+ };
+ 
+--- linux-2.6.22.1.orig/sound/pci/hda/patch_sigmatel.c
++++ linux-2.6.22.1/sound/pci/hda/patch_sigmatel.c
+@@ -44,6 +44,7 @@
+ 
+ enum {
+ 	STAC_9205_REF,
++	STAC_M43xx,
+ 	STAC_9205_MODELS
+ };
+ 
+@@ -59,11 +60,19 @@
+ 	STAC_D945_REF,
+ 	STAC_D945GTP3,
+ 	STAC_D945GTP5,
++	STAC_922X_DELL,
++	STAC_INTEL_MAC_V1,
++	STAC_INTEL_MAC_V2,
++	STAC_INTEL_MAC_V3,
++	STAC_INTEL_MAC_V4,
++	STAC_INTEL_MAC_V5,
++	/* for backward compitability */
+ 	STAC_MACMINI,
+ 	STAC_MACBOOK,
+ 	STAC_MACBOOK_PRO_V1,
+ 	STAC_MACBOOK_PRO_V2,
+ 	STAC_IMAC_INTEL,
++	STAC_IMAC_INTEL_20,
+ 	STAC_922X_MODELS
+ };
+ 
+@@ -210,7 +219,6 @@
+ 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 	0x0f, 0x14, 0x16, 0x17, 0x18,
+ 	0x21, 0x22,
+-	
+ };
+ 
+ static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol,
+@@ -326,8 +334,6 @@
+ };
+ 
+ static struct snd_kcontrol_new stac925x_mixer[] = {
+-	HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT),
+-	HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT),
+ 	{
+ 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ 		.name = "Input Source",
+@@ -549,44 +555,78 @@
+ 	0x02a19320, 0x40000100,
+ };
+ 
+-static unsigned int macbook_pro_v1_pin_configs[10] = {
+-	0x0321e230, 0x03a1e020, 0x9017e110, 0x01014010,
+-	0x01a19021, 0x0381e021, 0x1345e240, 0x13c5e22e,
+-	0x02a19320, 0x400000fb
++static unsigned int intel_mac_v1_pin_configs[10] = {
++	0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd,
++	0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240,
++	0x400000fc, 0x400000fb,
++};
++
++static unsigned int intel_mac_v2_pin_configs[10] = {
++	0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
++	0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa,
++	0x400000fc, 0x400000fb,
++};
++
++static unsigned int intel_mac_v3_pin_configs[10] = {
++	0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
++	0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240,
++	0x400000fc, 0x400000fb,
+ };
+ 
+-static unsigned int macbook_pro_v2_pin_configs[10] = {
+-	0x0221401f, 0x90a70120, 0x01813024, 0x01014010,
+-	0x400000fd, 0x01016011, 0x1345e240, 0x13c5e22e,
++static unsigned int intel_mac_v4_pin_configs[10] = {
++	0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
++	0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
+ 	0x400000fc, 0x400000fb,
+ };
+ 
+-static unsigned int imac_intel_pin_configs[10] = {
+-	0x0121e230, 0x90a70120, 0x9017e110, 0x400000fe,
+-	0x400000fd, 0x0181e021, 0x1145e040, 0x400000fa,
++static unsigned int intel_mac_v5_pin_configs[10] = {
++	0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
++	0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
+ 	0x400000fc, 0x400000fb,
+ };
+ 
++static unsigned int stac922x_dell_pin_configs[10] = {
++	0x0221121e, 0x408103ff, 0x02a1123e, 0x90100310,
++	0x408003f1, 0x0221122f, 0x03451340, 0x40c003f2,
++	0x50a003f3, 0x405003f4
++};
++
+ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
+ 	[STAC_D945_REF] = ref922x_pin_configs,
+ 	[STAC_D945GTP3] = d945gtp3_pin_configs,
+ 	[STAC_D945GTP5] = d945gtp5_pin_configs,
+-	[STAC_MACMINI] = macbook_pro_v1_pin_configs,
+-	[STAC_MACBOOK] = macbook_pro_v1_pin_configs,
+-	[STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs,
+-	[STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs,
+-	[STAC_IMAC_INTEL] = imac_intel_pin_configs,
++	[STAC_922X_DELL] = stac922x_dell_pin_configs,
++	[STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs,
++	[STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs,
++	[STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs,
++	[STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs,
++	[STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs,
++	/* for backward compitability */
++	[STAC_MACMINI] = intel_mac_v3_pin_configs,
++	[STAC_MACBOOK] = intel_mac_v5_pin_configs,
++	[STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs,
++	[STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs,
++	[STAC_IMAC_INTEL] = intel_mac_v2_pin_configs,
++	[STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs,
+ };
+ 
+ static const char *stac922x_models[STAC_922X_MODELS] = {
+ 	[STAC_D945_REF]	= "ref",
+ 	[STAC_D945GTP5]	= "5stack",
+ 	[STAC_D945GTP3]	= "3stack",
++	[STAC_922X_DELL] = "dell",
++	[STAC_INTEL_MAC_V1] = "intel-mac-v1",
++	[STAC_INTEL_MAC_V2] = "intel-mac-v2",
++	[STAC_INTEL_MAC_V3] = "intel-mac-v3",
++	[STAC_INTEL_MAC_V4] = "intel-mac-v4",
++	[STAC_INTEL_MAC_V5] = "intel-mac-v5",
++	/* for backward compitability */
+ 	[STAC_MACMINI]	= "macmini",
+ 	[STAC_MACBOOK]	= "macbook",
+ 	[STAC_MACBOOK_PRO_V1]	= "macbook-pro-v1",
+ 	[STAC_MACBOOK_PRO_V2]	= "macbook-pro",
+ 	[STAC_IMAC_INTEL] = "imac-intel",
++	[STAC_IMAC_INTEL_20] = "imac-intel-20",
+ };
+ 
+ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
+@@ -649,7 +689,10 @@
+ 	/* other systems  */
+ 	/* Apple Mac Mini (early 2006) */
+ 	SND_PCI_QUIRK(0x8384, 0x7680,
+-		      "Mac Mini", STAC_MACMINI),
++		      "Mac Mini", STAC_INTEL_MAC_V3),
++	/* Dell */
++	SND_PCI_QUIRK(0x1028, 0x01d7, "Dell XPS M1210", STAC_922X_DELL),
++
+ 	{} /* terminator */
+ };
+ 
+@@ -730,7 +773,8 @@
+ };
+ 
+ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
+-	ref9205_pin_configs,
++	[STAC_REF] = ref9205_pin_configs,
++	[STAC_M43xx] = NULL,
+ };
+ 
+ static const char *stac9205_models[STAC_9205_MODELS] = {
+@@ -741,6 +785,10 @@
+ 	/* SigmaTel reference board */
+ 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ 		      "DFI LanParty", STAC_9205_REF),
++	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8,
++		      "Dell Precision", STAC_M43xx),
++	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff,
++		      "Dell Precision", STAC_M43xx),
+ 	{} /* terminator */
+ };
+ 
+@@ -770,33 +818,56 @@
+ 	return 0;
+ }
+ 
++static void stac92xx_set_config_reg(struct hda_codec *codec,
++				    hda_nid_t pin_nid, unsigned int pin_config)
++{
++	int i;
++	snd_hda_codec_write(codec, pin_nid, 0,
++			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
++			    pin_config & 0x000000ff);
++	snd_hda_codec_write(codec, pin_nid, 0,
++			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
++			    (pin_config & 0x0000ff00) >> 8);
++	snd_hda_codec_write(codec, pin_nid, 0,
++			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
++			    (pin_config & 0x00ff0000) >> 16);
++	snd_hda_codec_write(codec, pin_nid, 0,
++			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
++			    pin_config >> 24);
++	i = snd_hda_codec_read(codec, pin_nid, 0,
++			       AC_VERB_GET_CONFIG_DEFAULT,
++			       0x00);	
++	snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n",
++		    pin_nid, i);
++}
++
+ static void stac92xx_set_config_regs(struct hda_codec *codec)
+ {
+ 	int i;
+ 	struct sigmatel_spec *spec = codec->spec;
+-	unsigned int pin_cfg;
+ 
+-	if (! spec->pin_nids || ! spec->pin_configs)
+-		return;
++ 	if (!spec->pin_configs)
++ 		return;
+ 
+-	for (i = 0; i < spec->num_pins; i++) {
+-		snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+-				    spec->pin_configs[i] & 0x000000ff);
+-		snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+-				    (spec->pin_configs[i] & 0x0000ff00) >> 8);
+-		snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+-				    (spec->pin_configs[i] & 0x00ff0000) >> 16);
+-		snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+-				    spec->pin_configs[i] >> 24);
+-		pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0,
+-					     AC_VERB_GET_CONFIG_DEFAULT,
+-					     0x00);	
+-		snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg);
+-	}
++	for (i = 0; i < spec->num_pins; i++)
++		stac92xx_set_config_reg(codec, spec->pin_nids[i],
++					spec->pin_configs[i]);
++}
++
++static void stac92xx_enable_gpio_mask(struct hda_codec *codec,
++				      int gpio_mask, int gpio_data)
++{
++	/* Configure GPIOx as output */
++	snd_hda_codec_write(codec, codec->afg, 0,
++			    AC_VERB_SET_GPIO_DIRECTION, gpio_mask);
++	/* Configure GPIOx as CMOS */
++	snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000);
++	/* Assert GPIOx */
++	snd_hda_codec_write(codec, codec->afg, 0,
++			    AC_VERB_SET_GPIO_DATA, gpio_data);
++	/* Enable GPIOx */
++	snd_hda_codec_write(codec, codec->afg, 0,
++			    AC_VERB_SET_GPIO_MASK, gpio_mask);
+ }
+ 
+ /*
+@@ -1168,7 +1239,7 @@
+  * and 9202/925x. For those, dac_nids[] must be hard-coded.
+  */
+ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
+-				       const struct auto_pin_cfg *cfg)
++				       struct auto_pin_cfg *cfg)
+ {
+ 	struct sigmatel_spec *spec = codec->spec;
+ 	int i, j, conn_len = 0; 
+@@ -1193,6 +1264,13 @@
+ 		}
+ 
+ 		if (j == conn_len) {
++			if (spec->multiout.num_dacs > 0) {
++				/* we have already working output pins,
++				 * so let's drop the broken ones again
++				 */
++				cfg->line_outs = spec->multiout.num_dacs;
++				break;
++			}
+ 			/* error out, no available DAC found */
+ 			snd_printk(KERN_ERR
+ 				   "%s: No available DAC for pin 0x%x\n",
+@@ -1334,7 +1412,15 @@
+ 			continue;
+ 		add_spec_dacs(spec, nid);
+ 	}
+-
++	for (i = 0; i < cfg->line_outs; i++) {
++		nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0,
++					AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
++		if (check_in_dac_nids(spec, nid))
++			nid = 0;
++		if (! nid)
++			continue;
++		add_spec_dacs(spec, nid);
++	}
+ 	for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) {
+ 		static const char *pfxs[] = {
+ 			"Speaker", "External Speaker", "Speaker2",
+@@ -1891,7 +1977,7 @@
+ 		return -ENOMEM;
+ 
+ 	codec->spec = spec;
+-	spec->num_pins = 8;
++	spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
+ 	spec->pin_nids = stac9200_pin_nids;
+ 	spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS,
+ 							stac9200_models,
+@@ -1941,7 +2027,7 @@
+ 		return -ENOMEM;
+ 
+ 	codec->spec = spec;
+-	spec->num_pins = 8;
++	spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
+ 	spec->pin_nids = stac925x_pin_nids;
+ 	spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS,
+ 							stac925x_models,
+@@ -2013,29 +2099,41 @@
+ 		return -ENOMEM;
+ 
+ 	codec->spec = spec;
+-	spec->num_pins = 10;
++	spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
+ 	spec->pin_nids = stac922x_pin_nids;
+ 	spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS,
+ 							stac922x_models,
+ 							stac922x_cfg_tbl);
+-	if (spec->board_config == STAC_MACMINI) {
++	if (spec->board_config == STAC_INTEL_MAC_V3) {
+ 		spec->gpio_mute = 1;
+ 		/* Intel Macs have all same PCI SSID, so we need to check
+ 		 * codec SSID to distinguish the exact models
+ 		 */
+ 		printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id);
+ 		switch (codec->subsystem_id) {
+-		case 0x106b0a00: /* MacBook First generatoin */
+-			spec->board_config = STAC_MACBOOK;
++
++		case 0x106b0800:
++			spec->board_config = STAC_INTEL_MAC_V1;
++			break;
++		case 0x106b0600:
++		case 0x106b0700:
++			spec->board_config = STAC_INTEL_MAC_V2;
+ 			break;
+-		case 0x106b0200: /* MacBook Pro first generation */
+-			spec->board_config = STAC_MACBOOK_PRO_V1;
++		case 0x106b0e00:
++		case 0x106b0f00:
++		case 0x106b1600:
++		case 0x106b1700:
++		case 0x106b0200:
++		case 0x106b1e00:
++			spec->board_config = STAC_INTEL_MAC_V3;
+ 			break;
+-		case 0x106b1e00: /* MacBook Pro second generation */
+-			spec->board_config = STAC_MACBOOK_PRO_V2;
++		case 0x106b1a00:
++		case 0x00000100:
++			spec->board_config = STAC_INTEL_MAC_V4;
+ 			break;
+-		case 0x106b0700: /* Intel-based iMac */
+-			spec->board_config = STAC_IMAC_INTEL;
++		case 0x106b0a00:
++		case 0x106b2200:
++			spec->board_config = STAC_INTEL_MAC_V5;
+ 			break;
+ 		}
+ 	}
+@@ -2082,6 +2180,13 @@
+ 
+ 	codec->patch_ops = stac92xx_patch_ops;
+ 
++	/* Fix Mux capture level; max to 2 */
++	snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT,
++				  (0 << AC_AMPCAP_OFFSET_SHIFT) |
++				  (2 << AC_AMPCAP_NUM_STEPS_SHIFT) |
++				  (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) |
++				  (0 << AC_AMPCAP_MUTE_SHIFT));
++
+ 	return 0;
+ }
+ 
+@@ -2095,7 +2200,7 @@
+ 		return -ENOMEM;
+ 
+ 	codec->spec = spec;
+-	spec->num_pins = 14;
++	spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
+ 	spec->pin_nids = stac927x_pin_nids;
+ 	spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS,
+ 							stac927x_models,
+@@ -2141,7 +2246,9 @@
+ 	}
+ 
+ 	spec->multiout.dac_nids = spec->dac_nids;
+-
++	/* GPIO0 High = Enable EAPD */
++	stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001);
++	
+ 	err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
+ 	if (!err) {
+ 		if (spec->board_config < 0) {
+@@ -2159,27 +2266,20 @@
+ 
+ 	codec->patch_ops = stac92xx_patch_ops;
+ 
+-	/* Fix Mux capture level; max to 2 */
+-	snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT,
+-				  (0 << AC_AMPCAP_OFFSET_SHIFT) |
+-				  (2 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+-				  (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+-				  (0 << AC_AMPCAP_MUTE_SHIFT));
+-
+ 	return 0;
+ }
+ 
+ static int patch_stac9205(struct hda_codec *codec)
+ {
+ 	struct sigmatel_spec *spec;
+-	int err;
++	int err, gpio_mask, gpio_data;
+ 
+ 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
+ 	if (spec == NULL)
+ 		return -ENOMEM;
+ 
+ 	codec->spec = spec;
+-	spec->num_pins = 14;
++	spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
+ 	spec->pin_nids = stac9205_pin_nids;
+ 	spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS,
+ 							stac9205_models,
+@@ -2209,19 +2309,21 @@
+ 	spec->mixer = stac9205_mixer;
+ 
+ 	spec->multiout.dac_nids = spec->dac_nids;
++	
++	if (spec->board_config == STAC_M43xx) {
++		/* Enable SPDIF in/out */
++		stac92xx_set_config_reg(codec, 0x1f, 0x01441030);
++		stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
++
++		gpio_mask = 0x00000007; /* GPIO0-2 */
++		/* GPIO0 High = EAPD, GPIO1 Low = DRM,
++		 * GPIO2 High = Headphone Mute
++		 */
++		gpio_data = 0x00000005;
++	} else
++		gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */
+ 
+-	/* Configure GPIO0 as EAPD output */
+-	snd_hda_codec_write(codec, codec->afg, 0,
+-			    AC_VERB_SET_GPIO_DIRECTION, 0x00000001);
+-	/* Configure GPIO0 as CMOS */
+-	snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000);
+-	/* Assert GPIO0 high */
+-	snd_hda_codec_write(codec, codec->afg, 0,
+-			    AC_VERB_SET_GPIO_DATA, 0x00000001);
+-	/* Enable GPIO0 */
+-	snd_hda_codec_write(codec, codec->afg, 0,
+-			    AC_VERB_SET_GPIO_MASK, 0x00000001);
+-
++	stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data);
+ 	err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
+ 	if (!err) {
+ 		if (spec->board_config < 0) {
+@@ -2256,8 +2358,8 @@
+ 	.num_items = 2,
+ 	.items = {
+ 		/* { "HP", 0x0 }, */
+-		{ "Line", 0x1 },
+-		{ "Mic", 0x2 },
++		{ "Mic Jack", 0x1 },
++		{ "Internal Mic", 0x2 },
+ 		{ "PCM", 0x3 },
+ 	}
+ };
+@@ -2268,7 +2370,7 @@
+ 	{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
+ 	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
+ 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */
++	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
+ 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
+ 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
+ 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
+@@ -2284,7 +2386,7 @@
+ 	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
+ /*	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */
+ 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */
++	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
+ 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
+ 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
+ /*	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */
+--- linux-2.6.22.1.orig/sound/pci/ice1712/revo.c
++++ linux-2.6.22.1/sound/pci/ice1712/revo.c
+@@ -186,7 +186,12 @@
+ #define AK_DAC(xname,xch) { .name = xname, .num_channels = xch }
+ 
+ static const struct snd_akm4xxx_dac_channel revo71_front[] = {
+-	AK_DAC("PCM Playback Volume", 2)
++	{
++		.name = "PCM Playback Volume",
++		.num_channels = 2,
++		/* front channels DAC supports muting */
++		.switch_name = "PCM Playback Switch",
++	},
+ };
+ 
+ static const struct snd_akm4xxx_dac_channel revo71_surround[] = {
+--- linux-2.6.22.1.orig/sound/pci/nm256/nm256.c
++++ linux-2.6.22.1/sound/pci/nm256/nm256.c
+@@ -1533,7 +1533,8 @@
+ 				printk(KERN_ERR "  force the driver to load by "
+ 				       "passing in the module parameter\n");
+ 				printk(KERN_ERR "    force_ac97=1\n");
+-				printk(KERN_ERR "  or try sb16 or cs423x drivers instead.\n");
++				printk(KERN_ERR "  or try sb16, opl3sa2, or "
++				       "cs423x drivers instead.\n");
+ 				err = -ENXIO;
+ 				goto __error;
+ 			}
+--- linux-2.6.22.1.orig/sound/pci/rme9652/rme9652.c
++++ linux-2.6.22.1/sound/pci/rme9652/rme9652.c
+@@ -406,7 +406,7 @@
+ 		} else if (!frag)
+ 			return 0;
+ 		offset -= rme9652->max_jitter;
+-		if (offset < 0)
++		if ((int)offset < 0)
+ 			offset += period_size * 2;
+ 	} else {
+ 		if (offset > period_size + rme9652->max_jitter) {
+--- linux-2.6.22.1.orig/sound/pci/via82xx.c
++++ linux-2.6.22.1/sound/pci/via82xx.c
+@@ -2098,7 +2098,7 @@
+ 		pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ 		if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
+ 			break;
+-		schedule_timeout_uninterruptible(1);
++		schedule_timeout(1);
+ 	} while (time_before(jiffies, end_time));
+ 
+ 	if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
+@@ -2117,7 +2117,7 @@
+ 			chip->ac97_secondary = 1;
+ 			goto __ac97_ok2;
+ 		}
+-		schedule_timeout_interruptible(1);
++		schedule_timeout(1);
+ 	} while (time_before(jiffies, end_time));
+ 	/* This is ok, the most of motherboards have only one codec */
+ 
+--- linux-2.6.22.1.orig/sound/pci/via82xx_modem.c
++++ linux-2.6.22.1/sound/pci/via82xx_modem.c
+@@ -983,7 +983,7 @@
+ 		pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ 		if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
+ 			break;
+-		schedule_timeout_uninterruptible(1);
++		schedule_timeout(1);
+ 	} while (time_before(jiffies, end_time));
+ 
+ 	if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
+@@ -1001,7 +1001,7 @@
+ 			chip->ac97_secondary = 1;
+ 			goto __ac97_ok2;
+ 		}
+-		schedule_timeout_interruptible(1);
++		schedule_timeout(1);
+ 	} while (time_before(jiffies, end_time));
+ 	/* This is ok, the most of motherboards have only one codec */
+ 
+--- linux-2.6.22.1.orig/sound/ppc/Kconfig
++++ linux-2.6.22.1/sound/ppc/Kconfig
+@@ -33,3 +33,23 @@
+ 	  option.
+ 
+ endmenu
++
++menu "ALSA PowerPC devices"
++	depends on SND!=n && ( PPC64 || PPC32 )
++
++config SND_PS3
++	tristate "PS3 Audio support"
++	depends on SND && PS3_PS3AV
++	select SND_PCM
++	default m
++	help
++	  Say Y here to include support for audio on the PS3
++
++	  To compile this driver as a module, choose M here: the module
++	  will be called snd_ps3.
++
++config SND_PS3_DEFAULT_START_DELAY
++	int "Startup delay time in ms"
++	depends on SND_PS3
++	default "2000"
++endmenu
+--- linux-2.6.22.1.orig/sound/ppc/Makefile
++++ linux-2.6.22.1/sound/ppc/Makefile
+@@ -6,4 +6,5 @@
+ snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o
+ 
+ # Toplevel Module Dependency
+-obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o
++obj-$(CONFIG_SND_POWERMAC)	+= snd-powermac.o
++obj-$(CONFIG_SND_PS3)		+= snd_ps3.o
+--- /dev/null
++++ linux-2.6.22.1/sound/ppc/snd_ps3.c
+@@ -0,0 +1,1125 @@
++/*
++ * Audio support for PS3
++ * Copyright (C) 2007 Sony Computer Entertainment Inc.
++ * All rights reserved.
++ * Copyright 2006, 2007 Sony Corporation
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License
++ * as published by the Free Software Foundation; version 2 of the Licence.
++ *
++ * This program is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
++ * GNU General Public License for more details.
++ *
++ * You should have received a copy of the GNU General Public License
++ * along with this program; if not, write to the Free Software
++ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
++ */
++
++#include <linux/init.h>
++#include <linux/slab.h>
++#include <linux/io.h>
++#include <linux/interrupt.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/initval.h>
++#include <sound/pcm.h>
++#include <sound/asound.h>
++#include <sound/memalloc.h>
++#include <sound/pcm_params.h>
++#include <sound/control.h>
++#include <linux/dmapool.h>
++#include <linux/dma-mapping.h>
++#include <asm/firmware.h>
++#include <linux/io.h>
++#include <asm/dma.h>
++#include <asm/lv1call.h>
++#include <asm/ps3.h>
++#include <asm/ps3av.h>
++
++#include "snd_ps3_reg.h"
++#include "snd_ps3.h"
++
++MODULE_LICENSE("GPL v2");
++MODULE_DESCRIPTION("PS3 sound driver");
++MODULE_AUTHOR("Sony Computer Entertainment Inc.");
++
++/* module  entries */
++static int __init snd_ps3_init(void);
++static void __exit snd_ps3_exit(void);
++
++/* ALSA snd driver ops */
++static int snd_ps3_pcm_open(struct snd_pcm_substream *substream);
++static int snd_ps3_pcm_close(struct snd_pcm_substream *substream);
++static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream);
++static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream,
++				 int cmd);
++static snd_pcm_uframes_t snd_ps3_pcm_pointer(struct snd_pcm_substream
++					     *substream);
++static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
++				 struct snd_pcm_hw_params *hw_params);
++static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream);
++
++
++/* ps3_system_bus_driver entries */
++static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev);
++static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev);
++
++/* address setup */
++static int snd_ps3_map_mmio(void);
++static void snd_ps3_unmap_mmio(void);
++static int snd_ps3_allocate_irq(void);
++static void snd_ps3_free_irq(void);
++static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start);
++
++/* interrupt handler */
++static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id);
++
++
++/* set sampling rate/format */
++static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream);
++/* take effect parameter change */
++static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card);
++/* initialize avsetting and take it effect */
++static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card);
++/* setup dma */
++static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
++			       enum snd_ps3_dma_filltype filltype);
++static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card);
++
++static dma_addr_t v_to_bus(struct snd_ps3_card_info *, void  *vaddr, int ch);
++
++
++module_init(snd_ps3_init);
++module_exit(snd_ps3_exit);
++
++/*
++ * global
++ */
++static struct snd_ps3_card_info the_card;
++
++static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY;
++
++module_param_named(start_delay, snd_ps3_start_delay, uint, 0644);
++MODULE_PARM_DESC(start_delay, "time to insert silent data in milisec");
++
++static int index = SNDRV_DEFAULT_IDX1;
++static char *id = SNDRV_DEFAULT_STR1;
++
++module_param(index, int, 0444);
++MODULE_PARM_DESC(index, "Index value for PS3 soundchip.");
++module_param(id, charp, 0444);
++MODULE_PARM_DESC(id, "ID string for PS3 soundchip.");
++
++
++/*
++ * PS3 audio register access
++ */
++static inline u32 read_reg(unsigned int reg)
++{
++	return in_be32(the_card.mapped_mmio_vaddr + reg);
++}
++static inline void write_reg(unsigned int reg, u32 val)
++{
++	out_be32(the_card.mapped_mmio_vaddr + reg, val);
++}
++static inline void update_reg(unsigned int reg, u32 or_val)
++{
++	u32 newval = read_reg(reg) | or_val;
++	write_reg(reg, newval);
++}
++static inline void update_mask_reg(unsigned int reg, u32 mask, u32 or_val)
++{
++	u32 newval = (read_reg(reg) & mask) | or_val;
++	write_reg(reg, newval);
++}
++
++/*
++ * ALSA defs
++ */
++const static struct snd_pcm_hardware snd_ps3_pcm_hw = {
++	.info = (SNDRV_PCM_INFO_MMAP |
++		 SNDRV_PCM_INFO_NONINTERLEAVED |
++		 SNDRV_PCM_INFO_MMAP_VALID),
++	.formats = (SNDRV_PCM_FMTBIT_S16_BE |
++		    SNDRV_PCM_FMTBIT_S24_BE),
++	.rates = (SNDRV_PCM_RATE_44100 |
++		  SNDRV_PCM_RATE_48000 |
++		  SNDRV_PCM_RATE_88200 |
++		  SNDRV_PCM_RATE_96000),
++	.rate_min = 44100,
++	.rate_max = 96000,
++
++	.channels_min = 2, /* stereo only */
++	.channels_max = 2,
++
++	.buffer_bytes_max = PS3_AUDIO_FIFO_SIZE * 64,
++
++	/* interrupt by four stages */
++	.period_bytes_min = PS3_AUDIO_FIFO_STAGE_SIZE * 4,
++	.period_bytes_max = PS3_AUDIO_FIFO_STAGE_SIZE * 4,
++
++	.periods_min = 16,
++	.periods_max = 32, /* buffer_size_max/ period_bytes_max */
++
++	.fifo_size = PS3_AUDIO_FIFO_SIZE
++};
++
++static struct snd_pcm_ops snd_ps3_pcm_spdif_ops =
++{
++	.open = snd_ps3_pcm_open,
++	.close = snd_ps3_pcm_close,
++	.prepare = snd_ps3_pcm_prepare,
++	.ioctl = snd_pcm_lib_ioctl,
++	.trigger = snd_ps3_pcm_trigger,
++	.pointer = snd_ps3_pcm_pointer,
++	.hw_params = snd_ps3_pcm_hw_params,
++	.hw_free = snd_ps3_pcm_hw_free
++};
++
++static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card,
++				   int count, int force_stop)
++{
++	int dma_ch, done, retries, stop_forced = 0;
++	uint32_t status;
++
++	for (dma_ch = 0; dma_ch < 8; dma_ch ++) {
++		retries = count;
++		do {
++			status = read_reg(PS3_AUDIO_KICK(dma_ch)) &
++				PS3_AUDIO_KICK_STATUS_MASK;
++			switch (status) {
++			case PS3_AUDIO_KICK_STATUS_DONE:
++			case PS3_AUDIO_KICK_STATUS_NOTIFY:
++			case PS3_AUDIO_KICK_STATUS_CLEAR:
++			case PS3_AUDIO_KICK_STATUS_ERROR:
++				done = 1;
++				break;
++			default:
++				done = 0;
++				udelay(10);
++			}
++		} while (!done && --retries);
++		if (!retries && force_stop) {
++			pr_info("%s: DMA ch %d is not stopped.",
++				__func__, dma_ch);
++			/* last resort. force to stop dma.
++			 *  NOTE: this cause DMA done interrupts
++			 */
++			update_reg(PS3_AUDIO_CONFIG, PS3_AUDIO_CONFIG_CLEAR);
++			stop_forced = 1;
++		}
++	}
++	return stop_forced;
++}
++
++/*
++ * wait for all dma is done.
++ * NOTE: caller should reset card->running before call.
++ *       If not, the interrupt handler will re-start DMA,
++ *       then DMA is never stopped.
++ */
++static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card)
++{
++	int stop_forced;
++	/*
++	 * wait for the last dma is done
++	 */
++
++	/*
++	 * expected maximum DMA done time is 5.7ms + something (DMA itself).
++	 * 5.7ms is from 16bit/sample 2ch 44.1Khz; the time next
++	 * DMA kick event would occur.
++	 */
++	stop_forced = snd_ps3_verify_dma_stop(card, 700, 1);
++
++	/*
++	 * clear outstanding interrupts.
++	 */
++	update_reg(PS3_AUDIO_INTR_0, 0);
++	update_reg(PS3_AUDIO_AX_IS, 0);
++
++	/*
++	 *revert CLEAR bit since it will not reset automatically after DMA stop
++	 */
++	if (stop_forced)
++		update_mask_reg(PS3_AUDIO_CONFIG, ~PS3_AUDIO_CONFIG_CLEAR, 0);
++	/* ensure the hardware sees changes */
++	wmb();
++}
++
++static void snd_ps3_kick_dma(struct snd_ps3_card_info *card)
++{
++
++	update_reg(PS3_AUDIO_KICK(0), PS3_AUDIO_KICK_REQUEST);
++	/* ensure the hardware sees the change */
++	wmb();
++}
++
++/*
++ * convert virtual addr to ioif bus addr.
++ */
++static dma_addr_t v_to_bus(struct snd_ps3_card_info *card,
++			   void * paddr,
++			   int ch)
++{
++	return card->dma_start_bus_addr[ch] +
++		(paddr - card->dma_start_vaddr[ch]);
++};
++
++
++/*
++ * increment ring buffer pointer.
++ * NOTE: caller must hold write spinlock
++ */
++static void snd_ps3_bump_buffer(struct snd_ps3_card_info *card,
++				enum snd_ps3_ch ch, size_t byte_count,
++				int stage)
++{
++	if (!stage)
++		card->dma_last_transfer_vaddr[ch] =
++			card->dma_next_transfer_vaddr[ch];
++	card->dma_next_transfer_vaddr[ch] += byte_count;
++	if ((card->dma_start_vaddr[ch] + (card->dma_buffer_size / 2)) <=
++	    card->dma_next_transfer_vaddr[ch]) {
++		card->dma_next_transfer_vaddr[ch] = card->dma_start_vaddr[ch];
++	}
++}
++/*
++ * setup dmac to send data to audio and attenuate samples on the ring buffer
++ */
++static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
++			       enum snd_ps3_dma_filltype filltype)
++{
++	/* this dmac does not support over 4G */
++	uint32_t dma_addr;
++	int fill_stages, dma_ch, stage;
++	enum snd_ps3_ch ch;
++	uint32_t ch0_kick_event = 0; /* initialize to mute gcc */
++	void *start_vaddr;
++	unsigned long irqsave;
++	int silent = 0;
++
++	switch (filltype) {
++	case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
++		silent = 1;
++		/* intentionally fall thru */
++	case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
++		ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
++		break;
++
++	case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
++		silent = 1;
++		/* intentionally fall thru */
++	case SND_PS3_DMA_FILLTYPE_RUNNING:
++		ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
++		break;
++	}
++
++	snd_ps3_verify_dma_stop(card, 700, 0);
++	fill_stages = 4;
++	spin_lock_irqsave(&card->dma_lock, irqsave);
++	for (ch = 0; ch < 2; ch++) {
++		start_vaddr = card->dma_next_transfer_vaddr[0];
++		for (stage = 0; stage < fill_stages; stage ++) {
++			dma_ch = stage * 2 + ch;
++			if (silent)
++				dma_addr = card->null_buffer_start_dma_addr;
++			else
++				dma_addr =
++				v_to_bus(card,
++					 card->dma_next_transfer_vaddr[ch],
++					 ch);
++
++			write_reg(PS3_AUDIO_SOURCE(dma_ch),
++				  (PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY |
++				   dma_addr));
++
++			/* dst: fixed to 3wire#0 */
++			if (ch == 0)
++				write_reg(PS3_AUDIO_DEST(dma_ch),
++					  (PS3_AUDIO_DEST_TARGET_AUDIOFIFO |
++					   PS3_AUDIO_AO_3W_LDATA(0)));
++			else
++				write_reg(PS3_AUDIO_DEST(dma_ch),
++					  (PS3_AUDIO_DEST_TARGET_AUDIOFIFO |
++					   PS3_AUDIO_AO_3W_RDATA(0)));
++
++			/* count always 1 DMA block (1/2 stage = 128 bytes) */
++			write_reg(PS3_AUDIO_DMASIZE(dma_ch), 0);
++			/* bump pointer if needed */
++			if (!silent)
++				snd_ps3_bump_buffer(card, ch,
++						    PS3_AUDIO_DMAC_BLOCK_SIZE,
++						    stage);
++
++			/* kick event  */
++			if (dma_ch == 0)
++				write_reg(PS3_AUDIO_KICK(dma_ch),
++					  ch0_kick_event);
++			else
++				write_reg(PS3_AUDIO_KICK(dma_ch),
++					  PS3_AUDIO_KICK_EVENT_AUDIO_DMA(dma_ch
++									 - 1) |
++					  PS3_AUDIO_KICK_REQUEST);
++		}
++	}
++	/* ensure the hardware sees the change */
++	wmb();
++	spin_unlock_irqrestore(&card->dma_lock, irqsave);
++
++	return 0;
++}
++
++/*
++ * audio mute on/off
++ * mute_on : 0 output enabled
++ *           1 mute
++ */
++static int snd_ps3_mute(int mute_on)
++{
++	return ps3av_audio_mute(mute_on);
++}
++
++/*
++ * PCM operators
++ */
++static int snd_ps3_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
++	int pcm_index;
++
++	pcm_index = substream->pcm->device;
++	/* to retrieve substream/runtime in interrupt handler */
++	card->substream = substream;
++
++	runtime->hw = snd_ps3_pcm_hw;
++
++	card->start_delay = snd_ps3_start_delay;
++
++	/* mute off */
++	snd_ps3_mute(0); /* this function sleep */
++
++	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
++				   PS3_AUDIO_FIFO_STAGE_SIZE * 4 * 2);
++	return 0;
++};
++
++static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
++				 struct snd_pcm_hw_params *hw_params)
++{
++	size_t size;
++
++	/* alloc transport buffer */
++	size = params_buffer_bytes(hw_params);
++	snd_pcm_lib_malloc_pages(substream, size);
++	return 0;
++};
++
++static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream,
++				  unsigned int delay_ms)
++{
++	int ret;
++	int rate ;
++
++	rate = substream->runtime->rate;
++	ret = snd_pcm_format_size(substream->runtime->format,
++				  rate * delay_ms / 1000)
++		* substream->runtime->channels;
++
++	pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n",
++		 __func__,
++		 delay_ms,
++		 rate,
++		 snd_pcm_format_size(substream->runtime->format, rate),
++		 rate * delay_ms / 1000,
++		 ret);
++
++	return ret;
++};
++
++static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
++	unsigned long irqsave;
++
++	if (!snd_ps3_set_avsetting(substream)) {
++		/* some parameter changed */
++		write_reg(PS3_AUDIO_AX_IE,
++			  PS3_AUDIO_AX_IE_ASOBEIE(0) |
++			  PS3_AUDIO_AX_IE_ASOBUIE(0));
++		/*
++		 * let SPDIF device re-lock with SPDIF signal,
++		 * start with some silence
++		 */
++		card->silent = snd_ps3_delay_to_bytes(substream,
++						      card->start_delay) /
++			(PS3_AUDIO_FIFO_STAGE_SIZE * 4); /* every 4 times */
++	}
++
++	/* restart ring buffer pointer */
++	spin_lock_irqsave(&card->dma_lock, irqsave);
++	{
++		card->dma_buffer_size = runtime->dma_bytes;
++
++		card->dma_last_transfer_vaddr[SND_PS3_CH_L] =
++			card->dma_next_transfer_vaddr[SND_PS3_CH_L] =
++			card->dma_start_vaddr[SND_PS3_CH_L] =
++			runtime->dma_area;
++		card->dma_start_bus_addr[SND_PS3_CH_L] = runtime->dma_addr;
++
++		card->dma_last_transfer_vaddr[SND_PS3_CH_R] =
++			card->dma_next_transfer_vaddr[SND_PS3_CH_R] =
++			card->dma_start_vaddr[SND_PS3_CH_R] =
++			runtime->dma_area + (runtime->dma_bytes / 2);
++		card->dma_start_bus_addr[SND_PS3_CH_R] =
++			runtime->dma_addr + (runtime->dma_bytes / 2);
++
++		pr_debug("%s: vaddr=%p bus=%#lx\n", __func__,
++			 card->dma_start_vaddr[SND_PS3_CH_L],
++			 card->dma_start_bus_addr[SND_PS3_CH_L]);
++
++	}
++	spin_unlock_irqrestore(&card->dma_lock, irqsave);
++
++	/* ensure the hardware sees the change */
++	mb();
++
++	return 0;
++};
++
++static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream,
++			       int cmd)
++{
++	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
++	int ret = 0;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		/* clear outstanding interrupts  */
++		update_reg(PS3_AUDIO_AX_IS, 0);
++
++		spin_lock(&card->dma_lock);
++		{
++			card->running = 1;
++		}
++		spin_unlock(&card->dma_lock);
++
++		snd_ps3_program_dma(card,
++				    SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
++		snd_ps3_kick_dma(card);
++		while (read_reg(PS3_AUDIO_KICK(7)) &
++		       PS3_AUDIO_KICK_STATUS_MASK) {
++			udelay(1);
++		}
++		snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
++		snd_ps3_kick_dma(card);
++		break;
++
++	case SNDRV_PCM_TRIGGER_STOP:
++		spin_lock(&card->dma_lock);
++		{
++			card->running = 0;
++		}
++		spin_unlock(&card->dma_lock);
++		snd_ps3_wait_for_dma_stop(card);
++		break;
++	default:
++		break;
++
++	}
++
++	return ret;
++};
++
++/*
++ * report current pointer
++ */
++static snd_pcm_uframes_t snd_ps3_pcm_pointer(
++	struct snd_pcm_substream *substream)
++{
++	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
++	size_t bytes;
++	snd_pcm_uframes_t ret;
++
++	spin_lock(&card->dma_lock);
++	{
++		bytes = (size_t)(card->dma_last_transfer_vaddr[SND_PS3_CH_L] -
++				 card->dma_start_vaddr[SND_PS3_CH_L]);
++	}
++	spin_unlock(&card->dma_lock);
++
++	ret = bytes_to_frames(substream->runtime, bytes * 2);
++
++	return ret;
++};
++
++static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream)
++{
++	int ret;
++	ret = snd_pcm_lib_free_pages(substream);
++	return ret;
++};
++
++static int snd_ps3_pcm_close(struct snd_pcm_substream *substream)
++{
++	/* mute on */
++	snd_ps3_mute(1);
++	return 0;
++};
++
++static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card)
++{
++	/*
++	 * avsetting driver seems to never change the followings
++	 * so, init them here once
++	 */
++
++	/* no dma interrupt needed */
++	write_reg(PS3_AUDIO_INTR_EN_0, 0);
++
++	/* use every 4 buffer empty interrupt */
++	update_mask_reg(PS3_AUDIO_AX_IC,
++			PS3_AUDIO_AX_IC_AASOIMD_MASK,
++			PS3_AUDIO_AX_IC_AASOIMD_EVERY4);
++
++	/* enable 3wire clocks */
++	update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
++			~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED |
++			  PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED),
++			0);
++	update_reg(PS3_AUDIO_AO_3WMCTRL,
++		   PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT);
++}
++
++/*
++ * av setting
++ * NOTE: calling this function may generate audio interrupt.
++ */
++static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card)
++{
++	int ret, retries, i;
++	pr_debug("%s: start\n", __func__);
++
++	ret = ps3av_set_audio_mode(card->avs.avs_audio_ch,
++				  card->avs.avs_audio_rate,
++				  card->avs.avs_audio_width,
++				  card->avs.avs_audio_format,
++				  card->avs.avs_audio_source);
++	/*
++	 * Reset the following unwanted settings:
++	 */
++
++	/* disable all 3wire buffers */
++	update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
++			~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) |
++			  PS3_AUDIO_AO_3WMCTRL_ASOEN(1) |
++			  PS3_AUDIO_AO_3WMCTRL_ASOEN(2) |
++			  PS3_AUDIO_AO_3WMCTRL_ASOEN(3)),
++			0);
++	wmb(); 	/* ensure the hardware sees the change */
++	/* wait for actually stopped */
++	retries = 1000;
++	while ((read_reg(PS3_AUDIO_AO_3WMCTRL) &
++		(PS3_AUDIO_AO_3WMCTRL_ASORUN(0) |
++		 PS3_AUDIO_AO_3WMCTRL_ASORUN(1) |
++		 PS3_AUDIO_AO_3WMCTRL_ASORUN(2) |
++		 PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) &&
++	       --retries) {
++		udelay(1);
++	}
++
++	/* reset buffer pointer */
++	for (i = 0; i < 4; i++) {
++		update_reg(PS3_AUDIO_AO_3WCTRL(i),
++			   PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET);
++		udelay(10);
++	}
++	wmb(); /* ensure the hardware actually start resetting */
++
++	/* enable 3wire#0 buffer */
++	update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0));
++
++
++	/* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */
++	update_mask_reg(PS3_AUDIO_AO_3WCTRL(0),
++			~PS3_AUDIO_AO_3WCTRL_ASODF,
++			PS3_AUDIO_AO_3WCTRL_ASODF_LSB);
++	update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0),
++			~PS3_AUDIO_AO_SPDCTRL_SPODF,
++			PS3_AUDIO_AO_SPDCTRL_SPODF_LSB);
++	/* ensure all the setting above is written back to register */
++	wmb();
++	/* avsetting driver altered AX_IE, caller must reset it if you want */
++	pr_debug("%s: end\n", __func__);
++	return ret;
++}
++
++static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card)
++{
++	int ret;
++	pr_debug("%s: start\n", __func__);
++	card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2;
++	card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
++	card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
++	card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM;
++	card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL;
++
++	ret = snd_ps3_change_avsetting(card);
++
++	snd_ps3_audio_fixup(card);
++
++	/* to start to generate SPDIF signal, fill data */
++	snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
++	snd_ps3_kick_dma(card);
++	pr_debug("%s: end\n", __func__);
++	return ret;
++}
++
++/*
++ *  set sampling rate according to the substream
++ */
++static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
++{
++	struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
++	struct snd_ps3_avsetting_info avs;
++
++	avs = card->avs;
++
++	pr_debug("%s: called freq=%d width=%d\n", __func__,
++		 substream->runtime->rate,
++		 snd_pcm_format_width(substream->runtime->format));
++
++	pr_debug("%s: before freq=%d width=%d\n", __func__,
++		 card->avs.avs_audio_rate, card->avs.avs_audio_width);
++
++	/* sample rate */
++	switch (substream->runtime->rate) {
++	case 44100:
++		avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K;
++		break;
++	case 48000:
++		avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
++		break;
++	case 88200:
++		avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K;
++		break;
++	case 96000:
++		avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K;
++		break;
++	default:
++		pr_info("%s: invalid rate %d\n", __func__,
++			substream->runtime->rate);
++		return 1;
++	}
++
++	/* width */
++	switch (snd_pcm_format_width(substream->runtime->format)) {
++	case 16:
++		avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
++		break;
++	case 24:
++		avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24;
++		break;
++	default:
++		pr_info("%s: invalid width %d\n", __func__,
++			snd_pcm_format_width(substream->runtime->format));
++		return 1;
++	}
++
++	if ((card->avs.avs_audio_width != avs.avs_audio_width) ||
++	    (card->avs.avs_audio_rate != avs.avs_audio_rate)) {
++		card->avs = avs;
++		snd_ps3_change_avsetting(card);
++
++		pr_debug("%s: after freq=%d width=%d\n", __func__,
++			 card->avs.avs_audio_rate, card->avs.avs_audio_width);
++
++		return 0;
++	} else
++		return 1;
++}
++
++
++
++static int snd_ps3_map_mmio(void)
++{
++	the_card.mapped_mmio_vaddr =
++		ioremap(the_card.ps3_dev->m_region->bus_addr,
++			the_card.ps3_dev->m_region->len);
++
++	if (!the_card.mapped_mmio_vaddr) {
++		pr_info("%s: ioremap 0 failed p=%#lx l=%#lx \n",
++		       __func__, the_card.ps3_dev->m_region->lpar_addr,
++		       the_card.ps3_dev->m_region->len);
++		return -ENXIO;
++	}
++
++	return 0;
++};
++
++static void snd_ps3_unmap_mmio(void)
++{
++	iounmap(the_card.mapped_mmio_vaddr);
++	the_card.mapped_mmio_vaddr = NULL;
++}
++
++static int snd_ps3_allocate_irq(void)
++{
++	int ret;
++	u64 lpar_addr, lpar_size;
++	u64 __iomem *mapped;
++
++	/* FIXME: move this to device_init (H/W probe) */
++
++	/* get irq outlet */
++	ret = lv1_gpu_device_map(1, &lpar_addr, &lpar_size);
++	if (ret) {
++		pr_info("%s: device map 1 failed %d\n", __func__,
++			ret);
++		return -ENXIO;
++	}
++
++	mapped = ioremap(lpar_addr, lpar_size);
++	if (!mapped) {
++		pr_info("%s: ioremap 1 failed \n", __func__);
++		return -ENXIO;
++	}
++
++	the_card.audio_irq_outlet = in_be64(mapped);
++
++	iounmap(mapped);
++	ret = lv1_gpu_device_unmap(1);
++	if (ret)
++		pr_info("%s: unmap 1 failed\n", __func__);
++
++	/* irq */
++	ret = ps3_irq_plug_setup(PS3_BINDING_CPU_ANY,
++				 the_card.audio_irq_outlet,
++				 &the_card.irq_no);
++	if (ret) {
++		pr_info("%s:ps3_alloc_irq failed (%d)\n", __func__, ret);
++		return ret;
++	}
++
++	ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
++			  SND_PS3_DRIVER_NAME, &the_card);
++	if (ret) {
++		pr_info("%s: request_irq failed (%d)\n", __func__, ret);
++		goto cleanup_irq;
++	}
++
++	return 0;
++
++ cleanup_irq:
++	ps3_irq_plug_destroy(the_card.irq_no);
++	return ret;
++};
++
++static void snd_ps3_free_irq(void)
++{
++	free_irq(the_card.irq_no, &the_card);
++	ps3_irq_plug_destroy(the_card.irq_no);
++}
++
++static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
++{
++	uint64_t val;
++	int ret;
++
++	val = (ioaddr_start & (0x0fUL << 32)) >> (32 - 20) |
++		(0x03UL << 24) |
++		(0x0fUL << 12) |
++		(PS3_AUDIO_IOID);
++
++	ret = lv1_gpu_attribute(0x100, 0x007, val, 0, 0);
++	if (ret)
++		pr_info("%s: gpu_attribute failed %d\n", __func__,
++			ret);
++}
++
++static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
++{
++	int ret;
++	u64 lpar_addr, lpar_size;
++
++	BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1));
++	BUG_ON(dev->match_id != PS3_MATCH_ID_SOUND);
++
++	the_card.ps3_dev = dev;
++
++	ret = ps3_open_hv_device(dev);
++
++	if (ret)
++		return -ENXIO;
++
++	/* setup MMIO */
++	ret = lv1_gpu_device_map(2, &lpar_addr, &lpar_size);
++	if (ret) {
++		pr_info("%s: device map 2 failed %d\n", __func__, ret);
++		goto clean_open;
++	}
++	ps3_mmio_region_init(dev, dev->m_region, lpar_addr, lpar_size,
++		PAGE_SHIFT);
++
++	ret = snd_ps3_map_mmio();
++	if (ret)
++		goto clean_dev_map;
++
++	/* setup DMA area */
++	ps3_dma_region_init(dev, dev->d_region,
++			    PAGE_SHIFT, /* use system page size */
++			    0, /* dma type; not used */
++			    NULL,
++			    _ALIGN_UP(SND_PS3_DMA_REGION_SIZE, PAGE_SIZE));
++	dev->d_region->ioid = PS3_AUDIO_IOID;
++
++	ret = ps3_dma_region_create(dev->d_region);
++	if (ret) {
++		pr_info("%s: region_create\n", __func__);
++		goto clean_mmio;
++	}
++
++	snd_ps3_audio_set_base_addr(dev->d_region->bus_addr);
++
++	/* CONFIG_SND_PS3_DEFAULT_START_DELAY */
++	the_card.start_delay = snd_ps3_start_delay;
++
++	/* irq */
++	if (snd_ps3_allocate_irq()) {
++		ret = -ENXIO;
++		goto clean_dma_region;
++	}
++
++	/* create card instance */
++	the_card.card = snd_card_new(index, id, THIS_MODULE, 0);
++	if (!the_card.card) {
++		ret = -ENXIO;
++		goto clean_irq;
++	}
++
++	strcpy(the_card.card->driver, "PS3");
++	strcpy(the_card.card->shortname, "PS3");
++	strcpy(the_card.card->longname, "PS3 sound");
++	/* create PCM devices instance */
++	/* NOTE:this driver works assuming pcm:substream = 1:1 */
++	ret = snd_pcm_new(the_card.card,
++			  "SPDIF",
++			  0, /* instance index, will be stored pcm.device*/
++			  1, /* output substream */
++			  0, /* input substream */
++			  &(the_card.pcm));
++	if (ret)
++		goto clean_card;
++
++	the_card.pcm->private_data = &the_card;
++	strcpy(the_card.pcm->name, "SPDIF");
++
++	/* set pcm ops */
++	snd_pcm_set_ops(the_card.pcm, SNDRV_PCM_STREAM_PLAYBACK,
++			&snd_ps3_pcm_spdif_ops);
++
++	the_card.pcm->info_flags = SNDRV_PCM_INFO_NONINTERLEAVED;
++	/* pre-alloc PCM DMA buffer*/
++	ret = snd_pcm_lib_preallocate_pages_for_all(the_card.pcm,
++					SNDRV_DMA_TYPE_DEV,
++					&dev->core,
++					SND_PS3_PCM_PREALLOC_SIZE,
++					SND_PS3_PCM_PREALLOC_SIZE);
++	if (ret < 0) {
++		pr_info("%s: prealloc failed\n", __func__);
++		goto clean_card;
++	}
++
++	/*
++	 * allocate null buffer
++	 * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2
++	 * PAGE_SIZE is enogh
++	 */
++	if (!(the_card.null_buffer_start_vaddr =
++	      dma_alloc_coherent(&the_card.ps3_dev->core,
++				 PAGE_SIZE,
++				 &the_card.null_buffer_start_dma_addr,
++				 GFP_KERNEL))) {
++		pr_info("%s: nullbuffer alloc failed\n", __func__);
++		goto clean_preallocate;
++	}
++	pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__,
++		 the_card.null_buffer_start_vaddr,
++		 the_card.null_buffer_start_dma_addr);
++	/* set default sample rate/word width */
++	snd_ps3_init_avsetting(&the_card);
++
++	/* register the card */
++	ret = snd_card_register(the_card.card);
++	if (ret < 0)
++		goto clean_dma_map;
++
++	pr_info("%s started. start_delay=%dms\n",
++		the_card.card->longname, the_card.start_delay);
++	return 0;
++
++clean_dma_map:
++	dma_free_coherent(&the_card.ps3_dev->core,
++			  PAGE_SIZE,
++			  the_card.null_buffer_start_vaddr,
++			  the_card.null_buffer_start_dma_addr);
++clean_preallocate:
++	snd_pcm_lib_preallocate_free_for_all(the_card.pcm);
++clean_card:
++	snd_card_free(the_card.card);
++clean_irq:
++	snd_ps3_free_irq();
++clean_dma_region:
++	ps3_dma_region_free(dev->d_region);
++clean_mmio:
++	snd_ps3_unmap_mmio();
++clean_dev_map:
++	lv1_gpu_device_unmap(2);
++clean_open:
++	ps3_close_hv_device(dev);
++	/*
++	 * there is no destructor function to pcm.
++	 * midlayer automatically releases if the card removed
++	 */
++	return ret;
++}; /* snd_ps3_probe */
++
++/* called when module removal */
++static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev)
++{
++	int ret;
++	pr_info("%s:start id=%d\n", __func__,  dev->match_id);
++	if (dev->match_id != PS3_MATCH_ID_SOUND)
++		return -ENXIO;
++
++	/*
++	 * ctl and preallocate buffer will be freed in
++	 * snd_card_free
++	 */
++	ret = snd_card_free(the_card.card);
++	if (ret)
++		pr_info("%s: ctl freecard=%d\n", __func__, ret);
++
++	dma_free_coherent(&dev->core,
++			  PAGE_SIZE,
++			  the_card.null_buffer_start_vaddr,
++			  the_card.null_buffer_start_dma_addr);
++
++	ps3_dma_region_free(dev->d_region);
++
++	snd_ps3_free_irq();
++	snd_ps3_unmap_mmio();
++
++	lv1_gpu_device_unmap(2);
++	ps3_close_hv_device(dev);
++	pr_info("%s:end id=%d\n", __func__, dev->match_id);
++	return 0;
++} /* snd_ps3_remove */
++
++static struct ps3_system_bus_driver snd_ps3_bus_driver_info = {
++	.match_id = PS3_MATCH_ID_SOUND,
++	.probe = snd_ps3_driver_probe,
++	.remove = snd_ps3_driver_remove,
++	.shutdown = snd_ps3_driver_remove,
++	.core = {
++		.name = SND_PS3_DRIVER_NAME,
++		.owner = THIS_MODULE,
++	},
++};
++
++
++/*
++ * Interrupt handler
++ */
++static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id)
++{
++
++	uint32_t port_intr;
++	int underflow_occured = 0;
++	struct snd_ps3_card_info *card = dev_id;
++
++	if (!card->running) {
++		update_reg(PS3_AUDIO_AX_IS, 0);
++		update_reg(PS3_AUDIO_INTR_0, 0);
++		return IRQ_HANDLED;
++	}
++
++	port_intr = read_reg(PS3_AUDIO_AX_IS);
++	/*
++	 *serial buffer empty detected (every 4 times),
++	 *program next dma and kick it
++	 */
++	if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) {
++		write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0));
++		if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
++			write_reg(PS3_AUDIO_AX_IS, port_intr);
++			underflow_occured = 1;
++		}
++		if (card->silent) {
++			/* we are still in silent time */
++			snd_ps3_program_dma(card,
++				(underflow_occured) ?
++				SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL :
++				SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
++			snd_ps3_kick_dma(card);
++			card->silent --;
++		} else {
++			snd_ps3_program_dma(card,
++				(underflow_occured) ?
++				SND_PS3_DMA_FILLTYPE_FIRSTFILL :
++				SND_PS3_DMA_FILLTYPE_RUNNING);
++			snd_ps3_kick_dma(card);
++			snd_pcm_period_elapsed(card->substream);
++		}
++	} else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
++		write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0));
++		/*
++		 * serial out underflow, but buffer empty not detected.
++		 * in this case, fill fifo with 0 to recover.  After
++		 * filling dummy data, serial automatically start to
++		 * consume them and then will generate normal buffer
++		 * empty interrupts.
++		 * If both buffer underflow and buffer empty are occured,
++		 * it is better to do nomal data transfer than empty one
++		 */
++		snd_ps3_program_dma(card,
++				    SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
++		snd_ps3_kick_dma(card);
++		snd_ps3_program_dma(card,
++				    SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
++		snd_ps3_kick_dma(card);
++	}
++	/* clear interrupt cause */
++	return IRQ_HANDLED;
++};
++
++/*
++ * module/subsystem initialize/terminate
++ */
++static int __init snd_ps3_init(void)
++{
++	int ret;
++
++	if (!firmware_has_feature(FW_FEATURE_PS3_LV1))
++		return -ENXIO;
++
++	memset(&the_card, 0, sizeof(the_card));
++	spin_lock_init(&the_card.dma_lock);
++
++	/* register systembus DRIVER, this calls our probe() func */
++	ret = ps3_system_bus_driver_register(&snd_ps3_bus_driver_info);
++
++	return ret;
++}
++
++static void __exit snd_ps3_exit(void)
++{
++	ps3_system_bus_driver_unregister(&snd_ps3_bus_driver_info);
++}
++
++MODULE_ALIAS(PS3_MODULE_ALIAS_SOUND);
+--- /dev/null
++++ linux-2.6.22.1/sound/ppc/snd_ps3.h
+@@ -0,0 +1,135 @@
++/*
++ * Audio support for PS3
++ * Copyright (C) 2007 Sony Computer Entertainment Inc.
++ * All rights reserved.
++ * Copyright 2006, 2007 Sony Corporation
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License
++ * as published by the Free Software Foundation; version 2 of the Licence.
++ *
++ * This program is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
++ * GNU General Public License for more details.
++ *
++ * You should have received a copy of the GNU General Public License
++ * along with this program; if not, write to the Free Software
++ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
++ */
++
++#if !defined(_SND_PS3_H_)
++#define _SND_PS3_H_
++
++#include <linux/irqreturn.h>
++
++#define SND_PS3_DRIVER_NAME "snd_ps3"
++
++enum snd_ps3_out_channel {
++	SND_PS3_OUT_SPDIF_0,
++	SND_PS3_OUT_SPDIF_1,
++	SND_PS3_OUT_SERIAL_0,
++	SND_PS3_OUT_DEVS
++};
++
++enum snd_ps3_dma_filltype {
++	SND_PS3_DMA_FILLTYPE_FIRSTFILL,
++	SND_PS3_DMA_FILLTYPE_RUNNING,
++	SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL,
++	SND_PS3_DMA_FILLTYPE_SILENT_RUNNING
++};
++
++enum snd_ps3_ch {
++	SND_PS3_CH_L = 0,
++	SND_PS3_CH_R = 1,
++	SND_PS3_CH_MAX = 2
++};
++
++struct snd_ps3_avsetting_info {
++	uint32_t avs_audio_ch;     /* fixed */
++	uint32_t avs_audio_rate;
++	uint32_t avs_audio_width;
++	uint32_t avs_audio_format; /* fixed */
++	uint32_t avs_audio_source; /* fixed */
++};
++/*
++ * PS3 audio 'card' instance
++ * there should be only ONE hardware.
++ */
++struct snd_ps3_card_info {
++	struct ps3_system_bus_device *ps3_dev;
++	struct snd_card *card;
++
++	struct snd_pcm *pcm;
++	struct snd_pcm_substream *substream;
++
++	/* hvc info */
++	u64 audio_lpar_addr;
++	u64 audio_lpar_size;
++
++	/* registers */
++	void __iomem *mapped_mmio_vaddr;
++
++	/* irq */
++	u64 audio_irq_outlet;
++	unsigned int irq_no;
++
++	/* remember avsetting */
++	struct snd_ps3_avsetting_info avs;
++
++	/* dma buffer management */
++	spinlock_t dma_lock;
++		/* dma_lock start */
++		void * dma_start_vaddr[2]; /* 0 for L, 1 for R */
++		dma_addr_t dma_start_bus_addr[2];
++		size_t dma_buffer_size;
++		void * dma_last_transfer_vaddr[2];
++		void * dma_next_transfer_vaddr[2];
++		int    silent;
++		/* dma_lock end */
++
++	int running;
++
++	/* null buffer */
++	void *null_buffer_start_vaddr;
++	dma_addr_t null_buffer_start_dma_addr;
++
++	/* start delay */
++	unsigned int start_delay;
++
++};
++
++
++/* PS3 audio DMAC block size in bytes */
++#define PS3_AUDIO_DMAC_BLOCK_SIZE (128)
++/* one stage (stereo)  of audio FIFO in bytes */
++#define PS3_AUDIO_FIFO_STAGE_SIZE (256)
++/* how many stages the fifo have */
++#define PS3_AUDIO_FIFO_STAGE_COUNT (8)
++/* fifo size 128 bytes * 8 stages * stereo (2ch) */
++#define PS3_AUDIO_FIFO_SIZE \
++	(PS3_AUDIO_FIFO_STAGE_SIZE * PS3_AUDIO_FIFO_STAGE_COUNT)
++
++/* PS3 audio DMAC max block count in one dma shot = 128 (0x80) blocks*/
++#define PS3_AUDIO_DMAC_MAX_BLOCKS  (PS3_AUDIO_DMASIZE_BLOCKS_MASK + 1)
++
++#define PS3_AUDIO_NORMAL_DMA_START_CH (0)
++#define PS3_AUDIO_NORMAL_DMA_COUNT    (8)
++#define PS3_AUDIO_NULL_DMA_START_CH \
++	(PS3_AUDIO_NORMAL_DMA_START_CH + PS3_AUDIO_NORMAL_DMA_COUNT)
++#define PS3_AUDIO_NULL_DMA_COUNT      (2)
++
++#define SND_PS3_MAX_VOL (0x0F)
++#define SND_PS3_MIN_VOL (0x00)
++#define SND_PS3_MIN_ATT SND_PS3_MIN_VOL
++#define SND_PS3_MAX_ATT SND_PS3_MAX_VOL
++
++#define SND_PS3_PCM_PREALLOC_SIZE \
++	(PS3_AUDIO_DMAC_BLOCK_SIZE * PS3_AUDIO_DMAC_MAX_BLOCKS * 4)
++
++#define SND_PS3_DMA_REGION_SIZE \
++	(SND_PS3_PCM_PREALLOC_SIZE + PAGE_SIZE)
++
++#define PS3_AUDIO_IOID       (1UL)
++
++#endif /* _SND_PS3_H_ */
+--- /dev/null
++++ linux-2.6.22.1/sound/ppc/snd_ps3_reg.h
+@@ -0,0 +1,891 @@
++/*
++ * Audio support for PS3
++ * Copyright (C) 2007 Sony Computer Entertainment Inc.
++ * Copyright 2006, 2007 Sony Corporation
++ * All rights reserved.
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of the GNU General Public License
++ * as published by the Free Software Foundation; version 2 of the License.
++ *
++ * This program is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
++ * GNU General Public License for more details.
++ *
++ * You should have received a copy of the GNU General Public License
++ * along with this program; if not, write to the Free Software
++ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
++ */
++
++/*
++ * interrupt / configure registers
++ */
++
++#define PS3_AUDIO_INTR_0                 (0x00000100)
++#define PS3_AUDIO_INTR_EN_0              (0x00000140)
++#define PS3_AUDIO_CONFIG                 (0x00000200)
++
++/*
++ * DMAC registers
++ * n:0..9
++ */
++#define PS3_AUDIO_DMAC_REGBASE(x)         (0x0000210 + 0x20 * (x))
++
++#define PS3_AUDIO_KICK(n)                 (PS3_AUDIO_DMAC_REGBASE(n) + 0x00)
++#define PS3_AUDIO_SOURCE(n)               (PS3_AUDIO_DMAC_REGBASE(n) + 0x04)
++#define PS3_AUDIO_DEST(n)                 (PS3_AUDIO_DMAC_REGBASE(n) + 0x08)
++#define PS3_AUDIO_DMASIZE(n)              (PS3_AUDIO_DMAC_REGBASE(n) + 0x0C)
++
++/*
++ * mute control
++ */
++#define PS3_AUDIO_AX_MCTRL                (0x00004000)
++#define PS3_AUDIO_AX_ISBP                 (0x00004004)
++#define PS3_AUDIO_AX_AOBP                 (0x00004008)
++#define PS3_AUDIO_AX_IC                   (0x00004010)
++#define PS3_AUDIO_AX_IE                   (0x00004014)
++#define PS3_AUDIO_AX_IS                   (0x00004018)
++
++/*
++ * three wire serial
++ * n:0..3
++ */
++#define PS3_AUDIO_AO_MCTRL                (0x00006000)
++#define PS3_AUDIO_AO_3WMCTRL              (0x00006004)
++
++#define PS3_AUDIO_AO_3WCTRL(n)            (0x00006200 + 0x200 * (n))
++
++/*
++ * S/PDIF
++ * n:0..1
++ * x:0..11
++ * y:0..5
++ */
++#define PS3_AUDIO_AO_SPD_REGBASE(n)       (0x00007200 + 0x200 * (n))
++
++#define PS3_AUDIO_AO_SPDCTRL(n) \
++	(PS3_AUDIO_AO_SPD_REGBASE(n) + 0x00)
++#define PS3_AUDIO_AO_SPDUB(n, x) \
++	(PS3_AUDIO_AO_SPD_REGBASE(n) + 0x04 + 0x04 * (x))
++#define PS3_AUDIO_AO_SPDCS(n, y) \
++	(PS3_AUDIO_AO_SPD_REGBASE(n) + 0x34 + 0x04 * (y))
++
++
++/*
++  PS3_AUDIO_INTR_0 register tells an interrupt handler which audio
++  DMA channel triggered the interrupt.  The interrupt status for a channel
++  can be cleared by writing a '1' to the corresponding bit.  A new interrupt
++  cannot be generated until the previous interrupt has been cleared.
++
++  Note that the status reported by PS3_AUDIO_INTR_0 is independent of the
++  value of PS3_AUDIO_INTR_EN_0.
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++#define PS3_AUDIO_INTR_0_CHAN(n)	(1 << ((n) * 2))
++#define PS3_AUDIO_INTR_0_CHAN9     PS3_AUDIO_INTR_0_CHAN(9)
++#define PS3_AUDIO_INTR_0_CHAN8     PS3_AUDIO_INTR_0_CHAN(8)
++#define PS3_AUDIO_INTR_0_CHAN7     PS3_AUDIO_INTR_0_CHAN(7)
++#define PS3_AUDIO_INTR_0_CHAN6     PS3_AUDIO_INTR_0_CHAN(6)
++#define PS3_AUDIO_INTR_0_CHAN5     PS3_AUDIO_INTR_0_CHAN(5)
++#define PS3_AUDIO_INTR_0_CHAN4     PS3_AUDIO_INTR_0_CHAN(4)
++#define PS3_AUDIO_INTR_0_CHAN3     PS3_AUDIO_INTR_0_CHAN(3)
++#define PS3_AUDIO_INTR_0_CHAN2     PS3_AUDIO_INTR_0_CHAN(2)
++#define PS3_AUDIO_INTR_0_CHAN1     PS3_AUDIO_INTR_0_CHAN(1)
++#define PS3_AUDIO_INTR_0_CHAN0     PS3_AUDIO_INTR_0_CHAN(0)
++
++/*
++  The PS3_AUDIO_INTR_EN_0 register specifies which DMA channels can generate
++  an interrupt to the PU.  Each bit of PS3_AUDIO_INTR_EN_0 is ANDed with the
++  corresponding bit in PS3_AUDIO_INTR_0.  The resulting bits are OR'd together
++  to generate the Audio interrupt.
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_EN_0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++  Bit assignments are same as PS3_AUDIO_INTR_0
++*/
++
++/*
++  PS3_AUDIO_CONFIG
++  31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 C|0 0 0 0 0 0 0 0| CONFIG
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++*/
++
++/* The CLEAR field cancels all pending transfers, and stops any running DMA
++   transfers.  Any interrupts associated with the canceled transfers
++   will occur as if the transfer had finished.
++   Since this bit is designed to recover from DMA related issues
++   which are caused by unpredictable situations, it is prefered to wait
++   for normal DMA transfer end without using this bit.
++*/
++#define PS3_AUDIO_CONFIG_CLEAR          (1 << 8)  /* RWIVF */
++
++/*
++  PS3_AUDIO_AX_MCTRL: Audio Port Mute Control Register
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|A|A|0 0 0 0 0 0 0|S|S|A|A|A|A| AX_MCTRL
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++/* 3 Wire Audio Serial Output Channel Mutes (0..3)  */
++#define PS3_AUDIO_AX_MCTRL_ASOMT(n)     (1 << (3 - (n)))  /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_ASO3MT       (1 << 0)          /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_ASO2MT       (1 << 1)          /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_ASO1MT       (1 << 2)          /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_ASO0MT       (1 << 3)          /* RWIVF */
++
++/* S/PDIF mutes (0,1)*/
++#define PS3_AUDIO_AX_MCTRL_SPOMT(n)     (1 << (5 - (n)))  /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_SPO1MT       (1 << 4)          /* RWIVF */
++#define PS3_AUDIO_AX_MCTRL_SPO0MT       (1 << 5)          /* RWIVF */
++
++/* All 3 Wire Serial Outputs Mute */
++#define PS3_AUDIO_AX_MCTRL_AASOMT       (1 << 13)         /* RWIVF */
++
++/* All S/PDIF Mute */
++#define PS3_AUDIO_AX_MCTRL_ASPOMT       (1 << 14)         /* RWIVF */
++
++/* All Audio Outputs Mute */
++#define PS3_AUDIO_AX_MCTRL_AAOMT        (1 << 15)         /* RWIVF */
++
++/*
++  S/PDIF Outputs Buffer Read/Write Pointer Register
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B|0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B| AX_ISBP
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++*/
++/*
++ S/PDIF Output Channel Read Buffer Numbers
++ Buffer number is  value of field.
++ Indicates current read access buffer ID from Audio Data
++ Transfer controller of S/PDIF Output
++*/
++
++#define PS3_AUDIO_AX_ISBP_SPOBRN_MASK(n) (0x7 << 4 * (1 - (n))) /* R-IUF */
++#define PS3_AUDIO_AX_ISBP_SPO1BRN_MASK		(0x7 << 0) /* R-IUF */
++#define PS3_AUDIO_AX_ISBP_SPO0BRN_MASK		(0x7 << 4) /* R-IUF */
++
++/*
++S/PDIF Output Channel Buffer Write Numbers
++Indicates current write access buffer ID from bus master.
++*/
++#define PS3_AUDIO_AX_ISBP_SPOBWN_MASK(n) (0x7 <<  4 * (5 - (n))) /* R-IUF */
++#define PS3_AUDIO_AX_ISBP_SPO1BWN_MASK		(0x7 << 16) /* R-IUF */
++#define PS3_AUDIO_AX_ISBP_SPO0BWN_MASK		(0x7 << 20) /* R-IUF */
++
++/*
++  3 Wire Audio Serial Outputs Buffer Read/Write
++  Pointer Register
++  Buffer number is  value of field
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B|0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B| AX_AOBP
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++/*
++3 Wire Audio Serial Output Channel Buffer Read Numbers
++Indicates current read access buffer Id from Audio Data Transfer
++Controller of 3 Wire Audio Serial Output Channels
++*/
++#define PS3_AUDIO_AX_AOBP_ASOBRN_MASK(n) (0x7 << 4 * (3 - (n))) /* R-IUF */
++
++#define PS3_AUDIO_AX_AOBP_ASO3BRN_MASK	(0x7 << 0) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO2BRN_MASK	(0x7 << 4) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO1BRN_MASK	(0x7 << 8) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO0BRN_MASK	(0x7 << 12) /* R-IUF */
++
++/*
++3 Wire Audio Serial Output Channel Buffer Write Numbers
++Indicates current write access buffer ID from bus master.
++*/
++#define PS3_AUDIO_AX_AOBP_ASOBWN_MASK(n) (0x7 << 4 * (7 - (n))) /* R-IUF */
++
++#define PS3_AUDIO_AX_AOBP_ASO3BWN_MASK        (0x7 << 16) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO2BWN_MASK        (0x7 << 20) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO1BWN_MASK        (0x7 << 24) /* R-IUF */
++#define PS3_AUDIO_AX_AOBP_ASO0BWN_MASK        (0x7 << 28) /* R-IUF */
++
++
++
++/*
++Audio Port Interrupt Condition Register
++For the fields in this register, the following values apply:
++0 = Interrupt is generated every interrupt event.
++1 = Interrupt is generated every 2 interrupt events.
++2 = Interrupt is generated every 4 interrupt events.
++3 = Reserved
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0|0 0|SPO|0 0|SPO|0 0|AAS|0 0 0 0 0 0 0 0 0 0 0 0| AX_IC
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++/*
++All 3-Wire Audio Serial Outputs Interrupt Mode
++Configures the Interrupt and Signal Notification
++condition of all 3-wire Audio Serial Outputs.
++*/
++#define PS3_AUDIO_AX_IC_AASOIMD_MASK          (0x3 << 12) /* RWIVF */
++#define PS3_AUDIO_AX_IC_AASOIMD_EVERY1        (0x0 << 12) /* RWI-V */
++#define PS3_AUDIO_AX_IC_AASOIMD_EVERY2        (0x1 << 12) /* RW--V */
++#define PS3_AUDIO_AX_IC_AASOIMD_EVERY4        (0x2 << 12) /* RW--V */
++
++/*
++S/PDIF Output Channel Interrupt Modes
++Configures the Interrupt and signal Notification
++conditions of S/PDIF output channels.
++*/
++#define PS3_AUDIO_AX_IC_SPO1IMD_MASK          (0x3 << 16) /* RWIVF */
++#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY1        (0x0 << 16) /* RWI-V */
++#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY2        (0x1 << 16) /* RW--V */
++#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY4        (0x2 << 16) /* RW--V */
++
++#define PS3_AUDIO_AX_IC_SPO0IMD_MASK          (0x3 << 20) /* RWIVF */
++#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY1        (0x0 << 20) /* RWI-V */
++#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY2        (0x1 << 20) /* RW--V */
++#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY4        (0x2 << 20) /* RW--V */
++
++/*
++Audio Port interrupt Enable Register
++Configures whether to enable or disable each Interrupt Generation.
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IE
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++*/
++
++/*
++3 Wire Audio Serial Output Channel Buffer Underflow
++Interrupt Enables
++Select enable/disable of Buffer Underflow Interrupts for
++3-Wire Audio Serial Output Channels
++DISABLED=Interrupt generation disabled.
++*/
++#define PS3_AUDIO_AX_IE_ASOBUIE(n)      (1 << (3 - (n))) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO3BUIE        (1 << 0) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO2BUIE        (1 << 1) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO1BUIE        (1 << 2) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO0BUIE        (1 << 3) /* RWIVF */
++
++/* S/PDIF Output Channel Buffer Underflow Interrupt Enables */
++
++#define PS3_AUDIO_AX_IE_SPOBUIE(n)      (1 << (7 - (n))) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO1BUIE        (1 << 6) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO0BUIE        (1 << 7) /* RWIVF */
++
++/* S/PDIF Output Channel One Block Transfer Completion Interrupt Enables */
++
++#define PS3_AUDIO_AX_IE_SPOBTCIE(n)     (1 << (11 - (n))) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO1BTCIE       (1 << 10) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO0BTCIE       (1 << 11) /* RWIVF */
++
++/* 3-Wire Audio Serial Output Channel Buffer Empty Interrupt Enables */
++
++#define PS3_AUDIO_AX_IE_ASOBEIE(n)      (1 << (19 - (n))) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO3BEIE        (1 << 16) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO2BEIE        (1 << 17) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO1BEIE        (1 << 18) /* RWIVF */
++#define PS3_AUDIO_AX_IE_ASO0BEIE        (1 << 19) /* RWIVF */
++
++/* S/PDIF Output Channel Buffer Empty Interrupt Enables */
++
++#define PS3_AUDIO_AX_IE_SPOBEIE(n)      (1 << (23 - (n))) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO1BEIE        (1 << 22) /* RWIVF */
++#define PS3_AUDIO_AX_IE_SPO0BEIE        (1 << 23) /* RWIVF */
++
++/*
++Audio Port Interrupt Status Register
++Indicates Interrupt status, which interrupt has occured, and can clear
++each interrupt in this register.
++Writing 1b to a field containing 1b clears field and de-asserts interrupt.
++Writing 0b to a field has no effect.
++Field vaules are the following:
++0 - Interrupt hasn't occured.
++1 - Interrupt has occured.
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IS
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++ Bit assignment are same as AX_IE
++*/
++
++/*
++Audio Output Master Control Register
++Configures Master Clock and other master Audio Output Settings
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0|SCKSE|0|SCKSE|  MR0  |  MR1  |MCL|MCL|0 0 0 0|0 0 0 0 0 0 0 0| AO_MCTRL
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++/*
++MCLK Output Control
++Controls mclko[1] output.
++0 - Disable output (fixed at High)
++1 - Output clock produced by clock selected
++with scksel1 by mr1
++2 - Reserved
++3 - Reserved
++*/
++
++#define PS3_AUDIO_AO_MCTRL_MCLKC1_MASK		(0x3 << 12) /* RWIVF */
++#define PS3_AUDIO_AO_MCTRL_MCLKC1_DISABLED	(0x0 << 12) /* RWI-V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC1_ENABLED	(0x1 << 12) /* RW--V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD2	(0x2 << 12) /* RW--V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD3	(0x3 << 12) /* RW--V */
++
++/*
++MCLK Output Control
++Controls mclko[0] output.
++0 - Disable output (fixed at High)
++1 - Output clock produced by clock selected
++with SCKSEL0 by MR0
++2 - Reserved
++3 - Reserved
++*/
++#define PS3_AUDIO_AO_MCTRL_MCLKC0_MASK		(0x3 << 14) /* RWIVF */
++#define PS3_AUDIO_AO_MCTRL_MCLKC0_DISABLED	(0x0 << 14) /* RWI-V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC0_ENABLED	(0x1 << 14) /* RW--V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD2	(0x2 << 14) /* RW--V */
++#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD3	(0x3 << 14) /* RW--V */
++/*
++Master Clock Rate 1
++Sets the divide ration of Master Clock1 (clock output from
++mclko[1] for the input clock selected by scksel1.
++*/
++#define PS3_AUDIO_AO_MCTRL_MR1_MASK	(0xf << 16)
++#define PS3_AUDIO_AO_MCTRL_MR1_DEFAULT	(0x0 << 16) /* RWI-V */
++/*
++Master Clock Rate 0
++Sets the divide ratio of Master Clock0 (clock output from
++mclko[0] for the input clock selected by scksel0).
++*/
++#define PS3_AUDIO_AO_MCTRL_MR0_MASK	(0xf << 20) /* RWIVF */
++#define PS3_AUDIO_AO_MCTRL_MR0_DEFAULT	(0x0 << 20) /* RWI-V */
++/*
++System Clock Select 0/1
++Selects the system clock to be used as Master Clock 0/1
++Input the system clock that is appropriate for the sampling
++rate.
++*/
++#define PS3_AUDIO_AO_MCTRL_SCKSEL1_MASK		(0x7 << 24) /* RWIVF */
++#define PS3_AUDIO_AO_MCTRL_SCKSEL1_DEFAULT	(0x2 << 24) /* RWI-V */
++
++#define PS3_AUDIO_AO_MCTRL_SCKSEL0_MASK		(0x7 << 28) /* RWIVF */
++#define PS3_AUDIO_AO_MCTRL_SCKSEL0_DEFAULT	(0x2 << 28) /* RWI-V */
++
++
++/*
++3-Wire Audio Output Master Control Register
++Configures clock, 3-Wire Audio Serial Output Enable, and
++other 3-Wire Audio Serial Output Master Settings
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |A|A|A|A|0 0 0|A| ASOSR |0 0 0 0|A|A|A|A|A|A|0|1|0 0 0 0 0 0 0 0| AO_3WMCTRL
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++
++/*
++LRCKO Polarity
++0 - Reserved
++1 - default
++*/
++#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK 		(1 << 8) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT	(1 << 8) /* RW--V */
++
++/* LRCK Output Disable */
++
++#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD		(1 << 10) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_ENABLED	(0 << 10) /* RW--V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED	(1 << 10) /* RWI-V */
++
++/* Bit Clock Output Disable */
++
++#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD		(1 << 11) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_ENABLED	(0 << 11) /* RW--V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED	(1 << 11) /* RWI-V */
++
++/*
++3-Wire Audio Serial Output Channel 0-3 Operational
++Status.  Each bit becomes 1 after each 3-Wire Audio
++Serial Output Channel N is in action by setting 1 to
++asoen.
++Each bit becomes 0 after each 3-Wire Audio Serial Output
++Channel N is out of action by setting 0 to asoen.
++*/
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN(n)		(1 << (15 - (n))) /* R-IVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(n)	(0 << (15 - (n))) /* R-I-V */
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(n)	(1 << (15 - (n))) /* R---V */
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN0		\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN(0)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_STOPPED	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(0)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_RUNNING	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(0)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN1		\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN(1)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_STOPPED	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(1)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_RUNNING	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(1)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN2		\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN(2)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_STOPPED	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(2)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_RUNNING	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(2)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN3		\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN(3)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_STOPPED	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(3)
++#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_RUNNING	\
++	PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(3)
++
++/*
++Sampling Rate
++Specifies the divide ratio of the bit clock (clock output
++from bclko) used by the 3-wire Audio Output Clock, whcih
++is applied to the master clock selected by mcksel.
++Data output is synchronized with this clock.
++*/
++#define PS3_AUDIO_AO_3WMCTRL_ASOSR_MASK		(0xf << 20) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV2		(0x1 << 20) /* RWI-V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV4		(0x2 << 20) /* RW--V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV8		(0x4 << 20) /* RW--V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV12	(0x6 << 20) /* RW--V */
++
++/*
++Master Clock Select
++0 - Master Clock 0
++1 - Master Clock 1
++*/
++#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL		(1 << 24) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK0	(0 << 24) /* RWI-V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK1	(1 << 24) /* RW--V */
++
++/*
++Enables and disables 4ch 3-Wire Audio Serial Output
++operation.  Each Bit from 0 to 3 corresponds to an
++output channel, which means that each output channel
++can be enabled or disabled individually.  When
++multiple channels are enabled at the same time, output
++operations are performed in synchronization.
++Bit 0 - Output Channel 0 (SDOUT[0])
++Bit 1 - Output Channel 1 (SDOUT[1])
++Bit 2 - Output Channel 2 (SDOUT[2])
++Bit 3 - Output Channel 3 (SDOUT[3])
++*/
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN(n)		(1 << (31 - (n))) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(n)	(0 << (31 - (n))) /* RWI-V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(n)	(1 << (31 - (n))) /* RW--V */
++
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN0 \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN(0) /* RWIVF */
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_DISABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(0) /* RWI-V */
++#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_ENABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(0) /* RW--V */
++#define PS3_AUDIO_A1_3WMCTRL_ASOEN0 \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN(1) /* RWIVF */
++#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_DISABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(1) /* RWI-V */
++#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_ENABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(1) /* RW--V */
++#define PS3_AUDIO_A2_3WMCTRL_ASOEN0 \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN(2) /* RWIVF */
++#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_DISABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(2) /* RWI-V */
++#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_ENABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(2) /* RW--V */
++#define PS3_AUDIO_A3_3WMCTRL_ASOEN0 \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN(3) /* RWIVF */
++#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_DISABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(3) /* RWI-V */
++#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_ENABLED \
++	PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(3) /* RW--V */
++
++/*
++3-Wire Audio Serial output Channel 0-3 Control Register
++Configures settings for 3-Wire Serial Audio Output Channel 0-3
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|0 0 0 0|A|0|ASO|0 0 0|0|0|0|0|0| AO_3WCTRL
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++*/
++/*
++Data Bit Mode
++Specifies the number of data bits
++0 - 16 bits
++1 - reserved
++2 - 20 bits
++3 - 24 bits
++*/
++#define PS3_AUDIO_AO_3WCTRL_ASODB_MASK	(0x3 << 8) /* RWIVF */
++#define PS3_AUDIO_AO_3WCTRL_ASODB_16BIT	(0x0 << 8) /* RWI-V */
++#define PS3_AUDIO_AO_3WCTRL_ASODB_RESVD	(0x1 << 8) /* RWI-V */
++#define PS3_AUDIO_AO_3WCTRL_ASODB_20BIT	(0x2 << 8) /* RW--V */
++#define PS3_AUDIO_AO_3WCTRL_ASODB_24BIT	(0x3 << 8) /* RW--V */
++/*
++Data Format Mode
++Specifies the data format where (LSB side or MSB) the data(in 20 bit
++or 24 bit resolution mode) is put in a 32 bit field.
++0 - Data put on LSB side
++1 - Data put on MSB side
++*/
++#define PS3_AUDIO_AO_3WCTRL_ASODF 	(1 << 11) /* RWIVF */
++#define PS3_AUDIO_AO_3WCTRL_ASODF_LSB	(0 << 11) /* RWI-V */
++#define PS3_AUDIO_AO_3WCTRL_ASODF_MSB	(1 << 11) /* RW--V */
++/*
++Buffer Reset
++Performs buffer reset.  Writing 1 to this bit initializes the
++corresponding 3-Wire Audio Output buffers(both L and R).
++*/
++#define PS3_AUDIO_AO_3WCTRL_ASOBRST 		(1 << 16) /* CWIVF */
++#define PS3_AUDIO_AO_3WCTRL_ASOBRST_IDLE	(0 << 16) /* -WI-V */
++#define PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET	(1 << 16) /* -W--T */
++
++/*
++S/PDIF Audio Output Channel 0/1 Control Register
++Configures settings for S/PDIF Audio Output Channel 0/1.
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |S|0 0 0|S|0 0|S| SPOSR |0 0|SPO|0 0 0 0|S|0|SPO|0 0 0 0 0 0 0|S| AO_SPDCTRL
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++/*
++Buffer reset.  Writing 1 to this bit initializes the
++corresponding S/PDIF output buffer pointer.
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPOBRST		(1 << 0) /* CWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_IDLE	(0 << 0) /* -WI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_RESET	(1 << 0) /* -W--T */
++
++/*
++Data Bit Mode
++Specifies number of data bits
++0 - 16 bits
++1 - Reserved
++2 - 20 bits
++3 - 24 bits
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPODB_MASK		(0x3 << 8) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPODB_16BIT	(0x0 << 8) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPODB_RESVD	(0x1 << 8) /* RW--V */
++#define PS3_AUDIO_AO_SPDCTRL_SPODB_20BIT	(0x2 << 8) /* RW--V */
++#define PS3_AUDIO_AO_SPDCTRL_SPODB_24BIT	(0x3 << 8) /* RW--V */
++/*
++Data format Mode
++Specifies the data format, where (LSB side or MSB)
++the data(in 20 or 24 bit resolution) is put in the
++32 bit field.
++0 - LSB Side
++1 - MSB Side
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPODF	(1 << 11) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPODF_LSB	(0 << 11) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPODF_MSB	(1 << 11) /* RW--V */
++/*
++Source Select
++Specifies the source of the S/PDIF output.  When 0, output
++operation is controlled by 3wen[0] of AO_3WMCTRL register.
++The SR must have the same setting as the a0_3wmctrl reg.
++0 - 3-Wire Audio OUT Ch0 Buffer
++1 - S/PDIF buffer
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPOSS_MASK		(0x3 << 16) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSS_3WEN		(0x0 << 16) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSS_SPDIF	(0x1 << 16) /* RW--V */
++/*
++Sampling Rate
++Specifies the divide ratio of the bit clock (clock output
++from bclko) used by the S/PDIF Output Clock, which
++is applied to the master clock selected by mcksel.
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPOSR		(0xf << 20) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV2		(0x1 << 20) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV4		(0x2 << 20) /* RW--V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV8		(0x4 << 20) /* RW--V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV12	(0x6 << 20) /* RW--V */
++/*
++Master Clock Select
++0 - Master Clock 0
++1 - Master Clock 1
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL		(1 << 24) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK0	(0 << 24) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK1	(1 << 24) /* RW--V */
++
++/*
++S/PDIF Output Channel Operational Status
++This bit becomes 1 after S/PDIF Output Channel is in
++action by setting 1 to spoen.  This bit becomes 0
++after S/PDIF Output Channel is out of action by setting
++0 to spoen.
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPORUN		(1 << 27) /* R-IVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPORUN_STOPPED	(0 << 27) /* R-I-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPORUN_RUNNING	(1 << 27) /* R---V */
++
++/*
++S/PDIF Audio Output Channel Output Enable
++Enables and disables output operation.  This bit is used
++only when sposs = 1
++*/
++#define PS3_AUDIO_AO_SPDCTRL_SPOEN		(1 << 31) /* RWIVF */
++#define PS3_AUDIO_AO_SPDCTRL_SPOEN_DISABLED	(0 << 31) /* RWI-V */
++#define PS3_AUDIO_AO_SPDCTRL_SPOEN_ENABLED	(1 << 31) /* RW--V */
++
++/*
++S/PDIF Audio Output Channel Channel Status
++Setting Registers.
++Configures channel status bit settings for each block
++(192 bits).
++Output is performed from the MSB(AO_SPDCS0 register bit 31).
++The same value is added for subframes within the same frame.
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |                             SPOCS                             | AO_SPDCS
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++
++S/PDIF Audio Output Channel User Bit Setting
++Configures user bit settings for each block (384 bits).
++Output is performed from the MSB(ao_spdub0 register bit 31).
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |                             SPOUB                             | AO_SPDUB
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++/*****************************************************************************
++ *
++ * DMAC register
++ *
++ *****************************************************************************/
++/*
++The PS3_AUDIO_KICK register is used to initiate a DMA transfer and monitor
++its status
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0|STATU|0 0 0|  EVENT  |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|R| KICK
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++/*
++The REQUEST field is written to ACTIVE to initiate a DMA request when EVENT
++occurs.
++It will return to the DONE state when the request is completed.
++The registers for a DMA channel should only be written if REQUEST is IDLE.
++*/
++
++#define PS3_AUDIO_KICK_REQUEST                (1 << 0) /* RWIVF */
++#define PS3_AUDIO_KICK_REQUEST_IDLE           (0 << 0) /* RWI-V */
++#define PS3_AUDIO_KICK_REQUEST_ACTIVE         (1 << 0) /* -W--T */
++
++/*
++ *The EVENT field is used to set the event in which
++ *the DMA request becomes active.
++ */
++#define PS3_AUDIO_KICK_EVENT_MASK             (0x1f << 16) /* RWIVF */
++#define PS3_AUDIO_KICK_EVENT_ALWAYS           (0x00 << 16) /* RWI-V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY (0x01 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_UNDERFLOW	(0x02 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_EMPTY		(0x03 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_UNDERFLOW	(0x04 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_EMPTY		(0x05 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_UNDERFLOW	(0x06 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_EMPTY		(0x07 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_UNDERFLOW	(0x08 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF0_BLOCKTRANSFERCOMPLETE \
++	(0x09 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF0_UNDERFLOW		(0x0A << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF0_EMPTY		(0x0B << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF1_BLOCKTRANSFERCOMPLETE \
++	(0x0C << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF1_UNDERFLOW		(0x0D << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_SPDIF1_EMPTY		(0x0E << 16) /* RW--V */
++
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA(n) \
++	((0x13 + (n)) << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA0         (0x13 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA1         (0x14 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA2         (0x15 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA3         (0x16 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA4         (0x17 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA5         (0x18 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA6         (0x19 << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA7         (0x1A << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA8         (0x1B << 16) /* RW--V */
++#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA9         (0x1C << 16) /* RW--V */
++
++/*
++The STATUS field can be used to monitor the progress of a DMA request.
++DONE indicates the previous request has completed.
++EVENT indicates that the DMA engine is waiting for the EVENT to occur.
++PENDING indicates that the DMA engine has not started processing this
++request, but the EVENT has occured.
++DMA indicates that the data transfer is in progress.
++NOTIFY indicates that the notifier signalling end of transfer is being written.
++CLEAR indicated that the previous transfer was cleared.
++ERROR indicates the previous transfer requested an unsupported
++source/destination combination.
++*/
++
++#define PS3_AUDIO_KICK_STATUS_MASK	(0x7 << 24) /* R-IVF */
++#define PS3_AUDIO_KICK_STATUS_DONE	(0x0 << 24) /* R-I-V */
++#define PS3_AUDIO_KICK_STATUS_EVENT	(0x1 << 24) /* R---V */
++#define PS3_AUDIO_KICK_STATUS_PENDING	(0x2 << 24) /* R---V */
++#define PS3_AUDIO_KICK_STATUS_DMA	(0x3 << 24) /* R---V */
++#define PS3_AUDIO_KICK_STATUS_NOTIFY	(0x4 << 24) /* R---V */
++#define PS3_AUDIO_KICK_STATUS_CLEAR	(0x5 << 24) /* R---V */
++#define PS3_AUDIO_KICK_STATUS_ERROR	(0x6 << 24) /* R---V */
++
++/*
++The PS3_AUDIO_SOURCE register specifies the source address for transfers.
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |                      START                      |0 0 0 0 0|TAR| SOURCE
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++/*
++The Audio DMA engine uses 128-byte transfers, thus the address must be aligned
++to a 128 byte boundary.  The low seven bits are assumed to be 0.
++*/
++
++#define PS3_AUDIO_SOURCE_START_MASK	(0x01FFFFFF << 7) /* RWIUF */
++
++/*
++The TARGET field specifies the memory space containing the source address.
++*/
++
++#define PS3_AUDIO_SOURCE_TARGET_MASK 		(3 << 0) /* RWIVF */
++#define PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY	(2 << 0) /* RW--V */
++
++/*
++The PS3_AUDIO_DEST register specifies the destination address for transfers.
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |                      START                      |0 0 0 0 0|TAR| DEST
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++/*
++The Audio DMA engine uses 128-byte transfers, thus the address must be aligned
++to a 128 byte boundary.  The low seven bits are assumed to be 0.
++*/
++
++#define PS3_AUDIO_DEST_START_MASK	(0x01FFFFFF << 7) /* RWIUF */
++
++/*
++The TARGET field specifies the memory space containing the destination address
++AUDIOFIFO = Audio WriteData FIFO,
++*/
++
++#define PS3_AUDIO_DEST_TARGET_MASK		(3 << 0) /* RWIVF */
++#define PS3_AUDIO_DEST_TARGET_AUDIOFIFO		(1 << 0) /* RW--V */
++
++/*
++PS3_AUDIO_DMASIZE specifies the number of 128-byte blocks + 1 to transfer.
++So a value of 0 means 128-bytes will get transfered.
++
++
++ 31            24 23           16 15            8 7             0
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|   BLOCKS    | DMASIZE
++ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
++*/
++
++
++#define PS3_AUDIO_DMASIZE_BLOCKS_MASK 	(0x7f << 0) /* RWIUF */
++
++/*
++ * source/destination address for internal fifos
++ */
++#define PS3_AUDIO_AO_3W_LDATA(n)	(0x1000 + (0x100 * (n)))
++#define PS3_AUDIO_AO_3W_RDATA(n)	(0x1080 + (0x100 * (n)))
++
++#define PS3_AUDIO_AO_SPD_DATA(n)	(0x2000 + (0x400 * (n)))
++
++
++/*
++ * field attiribute
++ *
++ *	Read
++ *	  ' ' = Other Information
++ *	  '-' = Field is part of a write-only register
++ *	  'C' = Value read is always the same, constant value line follows (C)
++ *	  'R' = Value is read
++ *
++ *	Write
++ *	  ' ' = Other Information
++ *	  '-' = Must not be written (D), value ignored when written (R,A,F)
++ *	  'W' = Can be written
++ *
++ *	Internal State
++ *	  ' ' = Other Information
++ *	  '-' = No internal state
++ *	  'X' = Internal state, initial value is unknown
++ *	  'I' = Internal state, initial value is known and follows (I)
++ *
++ *	Declaration/Size
++ *	  ' ' = Other Information
++ *	  '-' = Does Not Apply
++ *	  'V' = Type is void
++ *	  'U' = Type is unsigned integer
++ *	  'S' = Type is signed integer
++ *	  'F' = Type is IEEE floating point
++ *	  '1' = Byte size (008)
++ *	  '2' = Short size (016)
++ *	  '3' = Three byte size (024)
++ *	  '4' = Word size (032)
++ *	  '8' = Double size (064)
++ *
++ *	Define Indicator
++ *	  ' ' = Other Information
++ *	  'D' = Device
++ *	  'M' = Memory
++ *	  'R' = Register
++ *	  'A' = Array of Registers
++ *	  'F' = Field
++ *	  'V' = Value
++ *	  'T' = Task
++ */
++
+--- /dev/null
++++ linux-2.6.22.1/sound/sh/Kconfig
+@@ -0,0 +1,14 @@
++# ALSA SH drivers
++
++menu "SUPERH devices"
++	depends on SND!=n && SUPERH
++
++config SND_AICA
++	tristate "Dreamcast Yamaha AICA sound"
++	depends on SH_DREAMCAST && SND
++	select SND_PCM
++	help
++	  ALSA Sound driver for the SEGA Dreamcast console.
++
++endmenu
++
+--- /dev/null
++++ linux-2.6.22.1/sound/sh/Makefile
+@@ -0,0 +1,8 @@
++#
++# Makefile for ALSA
++#
++
++snd-aica-objs := aica.o
++
++# Toplevel Module Dependency
++obj-$(CONFIG_SND_AICA) += snd-aica.o
+--- /dev/null
++++ linux-2.6.22.1/sound/sh/aica.c
+@@ -0,0 +1,665 @@
++/*
++* This code is licenced under 
++* the General Public Licence
++* version 2
++*
++* Copyright Adrian McMenamin 2005, 2006, 2007
++* <adrian at mcmen.demon.co.uk>
++* Requires firmware (BSD licenced) available from:
++* http://linuxdc.cvs.sourceforge.net/linuxdc/linux-sh-dc/sound/oss/aica/firmware/
++* or the maintainer
++*
++* This program is free software; you can redistribute it and/or modify
++* it under the terms of version 2 of the GNU General Public License as published by
++* the Free Software Foundation.
++*
++* This program is distributed in the hope that it will be useful,
++* but WITHOUT ANY WARRANTY; without even the implied warranty of
++* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
++* GNU General Public License for more details.
++*
++* You should have received a copy of the GNU General Public License
++* along with this program; if not, write to the Free Software
++* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
++*
++*/
++
++#include <linux/init.h>
++#include <linux/jiffies.h>
++#include <linux/slab.h>
++#include <linux/time.h>
++#include <linux/wait.h>
++#include <linux/moduleparam.h>
++#include <linux/platform_device.h>
++#include <linux/firmware.h>
++#include <linux/timer.h>
++#include <linux/delay.h>
++#include <linux/workqueue.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/control.h>
++#include <sound/pcm.h>
++#include <sound/initval.h>
++#include <sound/info.h>
++#include <asm/io.h>
++#include <asm/dma.h>
++#include <asm/dreamcast/sysasic.h>
++#include "aica.h"
++
++MODULE_AUTHOR("Adrian McMenamin <adrian at mcmen.demon.co.uk>");
++MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver");
++MODULE_LICENSE("GPL");
++MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}");
++
++/* module parameters */
++#define CARD_NAME "AICA"
++static int index = -1;
++static char *id;
++static int enable = 1;
++module_param(index, int, 0444);
++MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
++module_param(id, charp, 0444);
++MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
++module_param(enable, bool, 0644);
++MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
++
++/* Use workqueue */
++static struct workqueue_struct *aica_queue;
++
++/* Simple platform device */
++static struct platform_device *pd;
++static struct resource aica_memory_space[2] = {
++	{
++	 .name = "AICA ARM CONTROL",
++	 .start = ARM_RESET_REGISTER,
++	 .flags = IORESOURCE_MEM,
++	 .end = ARM_RESET_REGISTER + 3,
++	 },
++	{
++	 .name = "AICA Sound RAM",
++	 .start = SPU_MEMORY_BASE,
++	 .flags = IORESOURCE_MEM,
++	 .end = SPU_MEMORY_BASE + 0x200000 - 1,
++	 },
++};
++
++/* SPU specific functions */
++/* spu_write_wait - wait for G2-SH FIFO to clear */
++static void spu_write_wait(void)
++{
++	int time_count;
++	time_count = 0;
++	while (1) {
++		if (!(readl(G2_FIFO) & 0x11))
++			break;
++		/* To ensure hardware failure doesn't wedge kernel */
++		time_count++;
++		if (time_count > 0x10000) {
++			snd_printk
++			    ("WARNING: G2 FIFO appears to be blocked.\n");
++			break;
++		}
++	}
++}
++
++/* spu_memset - write to memory in SPU address space */
++static void spu_memset(u32 toi, u32 what, int length)
++{
++	int i;
++	snd_assert(length % 4 == 0, return);
++	for (i = 0; i < length; i++) {
++		if (!(i % 8))
++			spu_write_wait();
++		writel(what, toi + SPU_MEMORY_BASE);
++		toi++;
++	}
++}
++
++/* spu_memload - write to SPU address space */
++static void spu_memload(u32 toi, void *from, int length)
++{
++	u32 *froml = from;
++	u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi);
++	int i;
++	u32 val;
++	length = DIV_ROUND_UP(length, 4);
++	spu_write_wait();
++	for (i = 0; i < length; i++) {
++		if (!(i % 8))
++			spu_write_wait();
++		val = *froml;
++		writel(val, to);
++		froml++;
++		to++;
++	}
++}
++
++/* spu_disable - set spu registers to stop sound output */
++static void spu_disable(void)
++{
++	int i;
++	u32 regval;
++	spu_write_wait();
++	regval = readl(ARM_RESET_REGISTER);
++	regval |= 1;
++	spu_write_wait();
++	writel(regval, ARM_RESET_REGISTER);
++	for (i = 0; i < 64; i++) {
++		spu_write_wait();
++		regval = readl(SPU_REGISTER_BASE + (i * 0x80));
++		regval = (regval & ~0x4000) | 0x8000;
++		spu_write_wait();
++		writel(regval, SPU_REGISTER_BASE + (i * 0x80));
++	}
++}
++
++/* spu_enable - set spu registers to enable sound output */
++static void spu_enable(void)
++{
++	u32 regval = readl(ARM_RESET_REGISTER);
++	regval &= ~1;
++	spu_write_wait();
++	writel(regval, ARM_RESET_REGISTER);
++}
++
++/* 
++ * Halt the sound processor, clear the memory,
++ * load some default ARM7 code, and then restart ARM7
++*/
++static void spu_reset(void)
++{
++	spu_disable();
++	spu_memset(0, 0, 0x200000 / 4);
++	/* Put ARM7 in endless loop */
++	ctrl_outl(0xea000002, SPU_MEMORY_BASE);
++	spu_enable();
++}
++
++/* aica_chn_start - write to spu to start playback */
++static void aica_chn_start(void)
++{
++	spu_write_wait();
++	writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT);
++}
++
++/* aica_chn_halt - write to spu to halt playback */
++static void aica_chn_halt(void)
++{
++	spu_write_wait();
++	writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT);
++}
++
++/* ALSA code below */
++static struct snd_pcm_hardware snd_pcm_aica_playback_hw = {
++	.info = (SNDRV_PCM_INFO_NONINTERLEAVED),
++	.formats =
++	    (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |
++	     SNDRV_PCM_FMTBIT_IMA_ADPCM),
++	.rates = SNDRV_PCM_RATE_8000_48000,
++	.rate_min = 8000,
++	.rate_max = 48000,
++	.channels_min = 1,
++	.channels_max = 2,
++	.buffer_bytes_max = AICA_BUFFER_SIZE,
++	.period_bytes_min = AICA_PERIOD_SIZE,
++	.period_bytes_max = AICA_PERIOD_SIZE,
++	.periods_min = AICA_PERIOD_NUMBER,
++	.periods_max = AICA_PERIOD_NUMBER,
++};
++
++static int aica_dma_transfer(int channels, int buffer_size,
++			     struct snd_pcm_substream *substream)
++{
++	int q, err, period_offset;
++	struct snd_card_aica *dreamcastcard;
++	struct snd_pcm_runtime *runtime;
++	err = 0;
++	dreamcastcard = substream->pcm->private_data;
++	period_offset = dreamcastcard->clicks;
++	period_offset %= (AICA_PERIOD_NUMBER / channels);
++	runtime = substream->runtime;
++	for (q = 0; q < channels; q++) {
++		err = dma_xfer(AICA_DMA_CHANNEL,
++			       (unsigned long) (runtime->dma_area +
++						(AICA_BUFFER_SIZE * q) /
++						channels +
++						AICA_PERIOD_SIZE *
++						period_offset),
++			       AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET +
++			       AICA_PERIOD_SIZE * period_offset,
++			       buffer_size / channels, AICA_DMA_MODE);
++		if (unlikely(err < 0))
++			break;
++		dma_wait_for_completion(AICA_DMA_CHANNEL);
++	}
++	return err;
++}
++
++static void startup_aica(struct snd_card_aica *dreamcastcard)
++{
++	spu_memload(AICA_CHANNEL0_CONTROL_OFFSET,
++		    dreamcastcard->channel, sizeof(struct aica_channel));
++	aica_chn_start();
++}
++
++static void run_spu_dma(struct work_struct *work)
++{
++	int buffer_size;
++	struct snd_pcm_runtime *runtime;
++	struct snd_card_aica *dreamcastcard;
++	dreamcastcard =
++	    container_of(work, struct snd_card_aica, spu_dma_work);
++	runtime = dreamcastcard->substream->runtime;
++	if (unlikely(dreamcastcard->dma_check == 0)) {
++		buffer_size =
++		    frames_to_bytes(runtime, runtime->buffer_size);
++		if (runtime->channels > 1)
++			dreamcastcard->channel->flags |= 0x01;
++		aica_dma_transfer(runtime->channels, buffer_size,
++				  dreamcastcard->substream);
++		startup_aica(dreamcastcard);
++		dreamcastcard->clicks =
++		    buffer_size / (AICA_PERIOD_SIZE * runtime->channels);
++		return;
++	} else {
++		aica_dma_transfer(runtime->channels,
++				  AICA_PERIOD_SIZE * runtime->channels,
++				  dreamcastcard->substream);
++		snd_pcm_period_elapsed(dreamcastcard->substream);
++		dreamcastcard->clicks++;
++		if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER))
++			dreamcastcard->clicks %= AICA_PERIOD_NUMBER;
++		mod_timer(&dreamcastcard->timer, jiffies + 1);
++	}
++}
++
++static void aica_period_elapsed(unsigned long timer_var)
++{
++	/*timer function - so cannot sleep */
++	int play_period;
++	struct snd_pcm_runtime *runtime;
++	struct snd_pcm_substream *substream;
++	struct snd_card_aica *dreamcastcard;
++	substream = (struct snd_pcm_substream *) timer_var;
++	runtime = substream->runtime;
++	dreamcastcard = substream->pcm->private_data;
++	/* Have we played out an additional period? */
++	play_period =
++	    frames_to_bytes(runtime,
++			    readl
++			    (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) /
++	    AICA_PERIOD_SIZE;
++	if (play_period == dreamcastcard->current_period) {
++		/* reschedule the timer */
++		mod_timer(&(dreamcastcard->timer), jiffies + 1);
++		return;
++	}
++	if (runtime->channels > 1)
++		dreamcastcard->current_period = play_period;
++	if (unlikely(dreamcastcard->dma_check == 0))
++		dreamcastcard->dma_check = 1;
++	queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
++}
++
++static void spu_begin_dma(struct snd_pcm_substream *substream)
++{
++	struct snd_card_aica *dreamcastcard;
++	struct snd_pcm_runtime *runtime;
++	runtime = substream->runtime;
++	dreamcastcard = substream->pcm->private_data;
++	/*get the queue to do the work */
++	queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
++	/* Timer may already be running */
++	if (unlikely(dreamcastcard->timer.data)) {
++		mod_timer(&dreamcastcard->timer, jiffies + 4);
++		return;
++	}
++	init_timer(&(dreamcastcard->timer));
++	dreamcastcard->timer.data = (unsigned long) substream;
++	dreamcastcard->timer.function = aica_period_elapsed;
++	dreamcastcard->timer.expires = jiffies + 4;
++	add_timer(&(dreamcastcard->timer));
++}
++
++static int snd_aicapcm_pcm_open(struct snd_pcm_substream
++				*substream)
++{
++	struct snd_pcm_runtime *runtime;
++	struct aica_channel *channel;
++	struct snd_card_aica *dreamcastcard;
++	if (!enable)
++		return -ENOENT;
++	dreamcastcard = substream->pcm->private_data;
++	channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL);
++	if (!channel)
++		return -ENOMEM;
++	/* set defaults for channel */
++	channel->sfmt = SM_8BIT;
++	channel->cmd = AICA_CMD_START;
++	channel->vol = dreamcastcard->master_volume;
++	channel->pan = 0x80;
++	channel->pos = 0;
++	channel->flags = 0;	/* default to mono */
++	dreamcastcard->channel = channel;
++	runtime = substream->runtime;
++	runtime->hw = snd_pcm_aica_playback_hw;
++	spu_enable();
++	dreamcastcard->clicks = 0;
++	dreamcastcard->current_period = 0;
++	dreamcastcard->dma_check = 0;
++	return 0;
++}
++
++static int snd_aicapcm_pcm_close(struct snd_pcm_substream
++				 *substream)
++{
++	struct snd_card_aica *dreamcastcard = substream->pcm->private_data;
++	flush_workqueue(aica_queue);
++	if (dreamcastcard->timer.data)
++		del_timer(&dreamcastcard->timer);
++	kfree(dreamcastcard->channel);
++	spu_disable();
++	return 0;
++}
++
++static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream
++				   *substream)
++{
++	/* Free the DMA buffer */
++	return snd_pcm_lib_free_pages(substream);
++}
++
++static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream
++				     *substream, struct snd_pcm_hw_params
++				     *hw_params)
++{
++	/* Allocate a DMA buffer using ALSA built-ins */
++	return
++	    snd_pcm_lib_malloc_pages(substream,
++				     params_buffer_bytes(hw_params));
++}
++
++static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream
++				   *substream)
++{
++	struct snd_card_aica *dreamcastcard = substream->pcm->private_data;
++	if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE)
++		dreamcastcard->channel->sfmt = SM_16BIT;
++	dreamcastcard->channel->freq = substream->runtime->rate;
++	dreamcastcard->substream = substream;
++	return 0;
++}
++
++static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream
++				   *substream, int cmd)
++{
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		spu_begin_dma(substream);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		aica_chn_halt();
++		break;
++	default:
++		return -EINVAL;
++	}
++	return 0;
++}
++
++static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream
++					     *substream)
++{
++	return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER);
++}
++
++static struct snd_pcm_ops snd_aicapcm_playback_ops = {
++	.open = snd_aicapcm_pcm_open,
++	.close = snd_aicapcm_pcm_close,
++	.ioctl = snd_pcm_lib_ioctl,
++	.hw_params = snd_aicapcm_pcm_hw_params,
++	.hw_free = snd_aicapcm_pcm_hw_free,
++	.prepare = snd_aicapcm_pcm_prepare,
++	.trigger = snd_aicapcm_pcm_trigger,
++	.pointer = snd_aicapcm_pcm_pointer,
++};
++
++/* TO DO: set up to handle more than one pcm instance */
++static int __init snd_aicapcmchip(struct snd_card_aica
++				  *dreamcastcard, int pcm_index)
++{
++	struct snd_pcm *pcm;
++	int err;
++	/* AICA has no capture ability */
++	err =
++	    snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0,
++			&pcm);
++	if (unlikely(err < 0))
++		return err;
++	pcm->private_data = dreamcastcard;
++	strcpy(pcm->name, "AICA PCM");
++	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
++			&snd_aicapcm_playback_ops);
++	/* Allocate the DMA buffers */
++	err =
++	    snd_pcm_lib_preallocate_pages_for_all(pcm,
++						  SNDRV_DMA_TYPE_CONTINUOUS,
++						  snd_dma_continuous_data
++						  (GFP_KERNEL),
++						  AICA_BUFFER_SIZE,
++						  AICA_BUFFER_SIZE);
++	return err;
++}
++
++/* Mixer controls */
++static int aica_pcmswitch_info(struct snd_kcontrol *kcontrol,
++			       struct snd_ctl_elem_info *uinfo)
++{
++	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
++	uinfo->count = 1;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = 1;
++	return 0;
++}
++
++static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol,
++			      struct snd_ctl_elem_value *ucontrol)
++{
++	ucontrol->value.integer.value[0] = 1;	/* TO DO: Fix me */
++	return 0;
++}
++
++static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol,
++			      struct snd_ctl_elem_value *ucontrol)
++{
++	if (ucontrol->value.integer.value[0] == 1)
++		return 0;	/* TO DO: Fix me */
++	else
++		aica_chn_halt();
++	return 0;
++}
++
++static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol,
++			       struct snd_ctl_elem_info *uinfo)
++{
++	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
++	uinfo->count = 1;
++	uinfo->value.integer.min = 0;
++	uinfo->value.integer.max = 0xFF;
++	return 0;
++}
++
++static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol,
++			      struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_card_aica *dreamcastcard;
++	dreamcastcard = kcontrol->private_data;
++	if (unlikely(!dreamcastcard->channel))
++		return -ETXTBSY;	/* we've not yet been set up */
++	ucontrol->value.integer.value[0] = dreamcastcard->channel->vol;
++	return 0;
++}
++
++static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol,
++			      struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_card_aica *dreamcastcard;
++	dreamcastcard = kcontrol->private_data;
++	if (unlikely(!dreamcastcard->channel))
++		return -ETXTBSY;
++	if (unlikely(dreamcastcard->channel->vol ==
++		     ucontrol->value.integer.value[0]))
++		return 0;
++	dreamcastcard->channel->vol = ucontrol->value.integer.value[0];
++	dreamcastcard->master_volume = ucontrol->value.integer.value[0];
++	spu_memload(AICA_CHANNEL0_CONTROL_OFFSET,
++		    dreamcastcard->channel, sizeof(struct aica_channel));
++	return 1;
++}
++
++static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = {
++	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++	.name = "PCM Playback Switch",
++	.index = 0,
++	.info = aica_pcmswitch_info,
++	.get = aica_pcmswitch_get,
++	.put = aica_pcmswitch_put
++};
++
++static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = {
++	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
++	.name = "PCM Playback Volume",
++	.index = 0,
++	.info = aica_pcmvolume_info,
++	.get = aica_pcmvolume_get,
++	.put = aica_pcmvolume_put
++};
++
++static int load_aica_firmware(void)
++{
++	int err;
++	const struct firmware *fw_entry;
++	spu_reset();
++	err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev);
++	if (unlikely(err))
++		return err;
++	/* write firware into memory */
++	spu_disable();
++	spu_memload(0, fw_entry->data, fw_entry->size);
++	spu_enable();
++	release_firmware(fw_entry);
++	return err;
++}
++
++static int __devinit add_aicamixer_controls(struct snd_card_aica
++					    *dreamcastcard)
++{
++	int err;
++	err = snd_ctl_add
++	    (dreamcastcard->card,
++	     snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard));
++	if (unlikely(err < 0))
++		return err;
++	err = snd_ctl_add
++	    (dreamcastcard->card,
++	     snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard));
++	if (unlikely(err < 0))
++		return err;
++	return 0;
++}
++
++static int snd_aica_remove(struct platform_device *devptr)
++{
++	struct snd_card_aica *dreamcastcard;
++	dreamcastcard = platform_get_drvdata(devptr);
++	if (unlikely(!dreamcastcard))
++		return -ENODEV;
++	snd_card_free(dreamcastcard->card);
++	kfree(dreamcastcard);
++	platform_set_drvdata(devptr, NULL);
++	return 0;
++}
++
++static int __init snd_aica_probe(struct platform_device *devptr)
++{
++	int err;
++	struct snd_card_aica *dreamcastcard;
++	dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL);
++	if (unlikely(!dreamcastcard))
++		return -ENOMEM;
++	dreamcastcard->card =
++	    snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0);
++	if (unlikely(!dreamcastcard->card)) {
++		kfree(dreamcastcard);
++		return -ENODEV;
++	}
++	strcpy(dreamcastcard->card->driver, "snd_aica");
++	strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER);
++	strcpy(dreamcastcard->card->longname,
++	       "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast");
++	/* Prepare to use the queue */
++	INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma);
++	/* Load the PCM 'chip' */
++	err = snd_aicapcmchip(dreamcastcard, 0);
++	if (unlikely(err < 0))
++		goto freedreamcast;
++	snd_card_set_dev(dreamcastcard->card, &devptr->dev);
++	dreamcastcard->timer.data = 0;
++	dreamcastcard->channel = NULL;
++	/* Add basic controls */
++	err = add_aicamixer_controls(dreamcastcard);
++	if (unlikely(err < 0))
++		goto freedreamcast;
++	/* Register the card with ALSA subsystem */
++	err = snd_card_register(dreamcastcard->card);
++	if (unlikely(err < 0))
++		goto freedreamcast;
++	platform_set_drvdata(devptr, dreamcastcard);
++	aica_queue = create_workqueue(CARD_NAME);
++	if (unlikely(!aica_queue))
++		goto freedreamcast;
++	snd_printk
++	    ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n");
++	return 0;
++      freedreamcast:
++	snd_card_free(dreamcastcard->card);
++	kfree(dreamcastcard);
++	return err;
++}
++
++static struct platform_driver snd_aica_driver = {
++	.probe = snd_aica_probe,
++	.remove = snd_aica_remove,
++	.driver = {
++		   .name = SND_AICA_DRIVER},
++};
++
++static int __init aica_init(void)
++{
++	int err;
++	err = platform_driver_register(&snd_aica_driver);
++	if (unlikely(err < 0))
++		return err;
++	pd = platform_device_register_simple(SND_AICA_DRIVER, -1,
++					     aica_memory_space, 2);
++	if (unlikely(IS_ERR(pd))) {
++		platform_driver_unregister(&snd_aica_driver);
++		return PTR_ERR(pd);
++	}
++	/* Load the firmware */
++	return load_aica_firmware();
++}
++
++static void __exit aica_exit(void)
++{
++	/* Destroy the aica kernel thread            *
++	 * being extra cautious to check if it exists*/
++	if (likely(aica_queue))
++		destroy_workqueue(aica_queue);
++	platform_device_unregister(pd);
++	platform_driver_unregister(&snd_aica_driver);
++	/* Kill any sound still playing and reset ARM7 to safe state */
++	spu_reset();
++}
++
++module_init(aica_init);
++module_exit(aica_exit);
+--- /dev/null
++++ linux-2.6.22.1/sound/sh/aica.h
+@@ -0,0 +1,81 @@
++/* aica.h
++ * Header file for ALSA driver for
++ * Sega Dreamcast Yamaha AICA sound
++ * Copyright Adrian McMenamin
++ * <adrian at mcmen.demon.co.uk>
++ * 2006
++ *
++ * This program is free software; you can redistribute it and/or modify
++ * it under the terms of version 2 of the GNU General Public License as published by
++ * the Free Software Foundation.
++ *
++ * This program is distributed in the hope that it will be useful,
++ * but WITHOUT ANY WARRANTY; without even the implied warranty of
++ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
++ * GNU General Public License for more details.
++ *
++ * You should have received a copy of the GNU General Public License
++ * along with this program; if not, write to the Free Software
++ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
++ *
++ */
++
++/* SPU memory and register constants etc */
++#define G2_FIFO 0xa05f688c
++#define SPU_MEMORY_BASE 0xA0800000
++#define ARM_RESET_REGISTER 0xA0702C00
++#define SPU_REGISTER_BASE 0xA0700000
++
++/* AICA channels stuff */
++#define AICA_CONTROL_POINT 0xA0810000
++#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008
++#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004
++
++/* Command values */
++#define AICA_CMD_KICK 0x80000000
++#define AICA_CMD_NONE 0
++#define AICA_CMD_START 1
++#define AICA_CMD_STOP 2
++#define AICA_CMD_VOL 3
++
++/* Sound modes */
++#define SM_8BIT		1
++#define SM_16BIT	0
++#define SM_ADPCM	2
++
++/* Buffer and period size */
++#define AICA_BUFFER_SIZE 0x8000
++#define AICA_PERIOD_SIZE 0x800
++#define AICA_PERIOD_NUMBER 16
++
++#define AICA_CHANNEL0_OFFSET 0x11000
++#define AICA_CHANNEL1_OFFSET 0x21000
++#define CHANNEL_OFFSET 0x10000
++
++#define AICA_DMA_CHANNEL 0
++#define AICA_DMA_MODE 5
++
++#define SND_AICA_DRIVER "AICA"
++
++struct aica_channel {
++	uint32_t cmd;		/* Command ID           */
++	uint32_t pos;		/* Sample position      */
++	uint32_t length;	/* Sample length        */
++	uint32_t freq;		/* Frequency            */
++	uint32_t vol;		/* Volume 0-255         */
++	uint32_t pan;		/* Pan 0-255            */
++	uint32_t sfmt;		/* Sound format         */
++	uint32_t flags;		/* Bit flags            */
++};
++
++struct snd_card_aica {
++	struct work_struct spu_dma_work;
++	struct snd_card *card;
++	struct aica_channel *channel;
++	struct snd_pcm_substream *substream;
++	int clicks;
++	int current_period;
++	struct timer_list timer;
++	int master_volume;
++	int dma_check;
++};
+--- linux-2.6.22.1.orig/sound/soc/Kconfig
++++ linux-2.6.22.1/sound/soc/Kconfig
+@@ -27,6 +27,7 @@
+ source "sound/soc/at91/Kconfig"
+ source "sound/soc/pxa/Kconfig"
+ source "sound/soc/s3c24xx/Kconfig"
++source "sound/soc/sh/Kconfig"
+ 
+ # Supported codecs
+ source "sound/soc/codecs/Kconfig"
+--- linux-2.6.22.1.orig/sound/soc/Makefile
++++ linux-2.6.22.1/sound/soc/Makefile
+@@ -1,4 +1,4 @@
+ snd-soc-core-objs := soc-core.o soc-dapm.o
+ 
+ obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
+-obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/
++obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/
+--- linux-2.6.22.1.orig/sound/soc/s3c24xx/Kconfig
++++ linux-2.6.22.1/sound/soc/s3c24xx/Kconfig
+@@ -1,6 +1,7 @@
+ config SND_S3C24XX_SOC
+ 	tristate "SoC Audio for the Samsung S3C24XX chips"
+ 	depends on ARCH_S3C2410 && SND_SOC
++	select SND_PCM
+ 	help
+ 	  Say Y or M if you want to add support for codecs attached to
+ 	  the S3C24XX AC97, I2S or SSP interface. You will also need
+@@ -8,3 +9,29 @@
+ 
+ config SND_S3C24XX_SOC_I2S
+ 	tristate
++
++config SND_S3C2443_SOC_AC97
++	tristate
++	select AC97_BUS
++	select SND_AC97_CODEC
++	select SND_SOC_AC97_BUS
++	
++config SND_S3C24XX_SOC_NEO1973_WM8753
++	tristate "SoC I2S Audio support for NEO1973 - WM8753"
++	depends on SND_S3C24XX_SOC && MACH_GTA01
++	select SND_S3C24XX_SOC_I2S
++	select SND_SOC_WM8753
++	help
++	  Say Y if you want to add support for SoC audio on smdk2440
++	  with the WM8753.
++
++config SND_S3C24XX_SOC_SMDK2443_WM9710
++	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
++	depends on SND_S3C24XX_SOC && MACH_SMDK2443
++	select SND_S3C2443_SOC_AC97
++	select SND_SOC_AC97_CODEC
++	help
++	  Say Y if you want to add support for SoC audio on smdk2443
++	  with the WM9710.
++
++
+--- linux-2.6.22.1.orig/sound/soc/s3c24xx/Makefile
++++ linux-2.6.22.1/sound/soc/s3c24xx/Makefile
+@@ -1,6 +1,15 @@
+ # S3c24XX Platform Support
+ snd-soc-s3c24xx-objs := s3c24xx-pcm.o
+ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
++snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
+ 
+ obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
+ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
++obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
++
++# S3C24XX Machine Support
++snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
++snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
++
++obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
++obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/s3c24xx/lm4857.h
+@@ -0,0 +1,32 @@
++/*
++ * lm4857.h  --  ALSA Soc Audio Layer
++ *
++ * Copyright 2007 Wolfson Microelectronics PLC.
++ * Author: Graeme Gregory
++ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    18th Jun 2007   Initial version.
++ */
++
++#ifndef LM4857_H_
++#define LM4857_H_
++
++/* The register offsets in the cache array */
++#define LM4857_MVOL 0
++#define LM4857_LVOL 1
++#define LM4857_RVOL 2
++#define LM4857_CTRL 3
++
++/* the shifts required to set these bits */
++#define LM4857_3D 5
++#define LM4857_WAKEUP 5
++#define LM4857_EPGAIN 4
++
++#endif /*LM4857_H_*/
++
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/s3c24xx/neo1973_wm8753.c
+@@ -0,0 +1,670 @@
++/*
++ * neo1973_wm8753.c  --  SoC audio for Neo1973
++ *
++ * Copyright 2007 Wolfson Microelectronics PLC.
++ * Author: Graeme Gregory
++ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    20th Jan 2007   Initial version.
++ *    05th Feb 2007   Rename all to Neo1973
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/timer.h>
++#include <linux/interrupt.h>
++#include <linux/platform_device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/mach-types.h>
++#include <asm/hardware/scoop.h>
++#include <asm/arch/regs-iis.h>
++#include <asm/arch/regs-clock.h>
++#include <asm/arch/regs-gpio.h>
++#include <asm/hardware.h>
++#include <asm/arch/audio.h>
++#include <asm/io.h>
++#include <asm/arch/spi-gpio.h>
++#include "../codecs/wm8753.h"
++#include "lm4857.h"
++#include "s3c24xx-pcm.h"
++#include "s3c24xx-i2s.h"
++
++/* define the scenarios */
++#define NEO_AUDIO_OFF			0
++#define NEO_GSM_CALL_AUDIO_HANDSET	1
++#define NEO_GSM_CALL_AUDIO_HEADSET	2
++#define NEO_GSM_CALL_AUDIO_BLUETOOTH	3
++#define NEO_STEREO_TO_SPEAKERS		4
++#define NEO_STEREO_TO_HEADPHONES	5
++#define NEO_CAPTURE_HANDSET		6
++#define NEO_CAPTURE_HEADSET		7
++#define NEO_CAPTURE_BLUETOOTH		8
++
++static struct snd_soc_machine neo1973;
++static struct i2c_client *i2c;
++
++static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++	unsigned int pll_out = 0, bclk = 0;
++	int ret = 0;
++	unsigned long iis_clkrate;
++
++	iis_clkrate = s3c24xx_i2s_get_clockrate();
++
++	switch (params_rate(params)) {
++	case 8000:
++	case 16000:
++		pll_out = 12288000;
++		break;
++	case 48000:
++		bclk = WM8753_BCLK_DIV_4;
++		pll_out = 12288000;
++		break;
++	case 96000:
++		bclk = WM8753_BCLK_DIV_2;
++		pll_out = 12288000;
++		break;
++	case 11025:
++		bclk = WM8753_BCLK_DIV_16;
++		pll_out = 11289600;
++		break;
++	case 22050:
++		bclk = WM8753_BCLK_DIV_8;
++		pll_out = 11289600;
++		break;
++	case 44100:
++		bclk = WM8753_BCLK_DIV_4;
++		pll_out = 11289600;
++		break;
++	case 88200:
++		bclk = WM8753_BCLK_DIV_2;
++		pll_out = 11289600;
++		break;
++	}
++
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai,
++		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
++		SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set cpu DAI configuration */
++	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
++		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
++		SND_SOC_DAIFMT_CBM_CFM);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set MCLK division for sample rate */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
++		S3C2410_IISMOD_32FS );
++	if (ret < 0)
++		return ret;
++
++	/* set codec BCLK division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
++	if (ret < 0)
++		return ret;
++
++	/* set prescaler division for sample rate */
++	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
++		S3C24XX_PRESCALE(4,4));
++	if (ret < 0)
++		return ret;
++
++	/* codec PLL input is PCLK/4 */
++	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1,
++		iis_clkrate / 4, pll_out);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++	/* disable the PLL */
++	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
++}
++
++/*
++ * Neo1973 WM8753 HiFi DAI opserations.
++ */
++static struct snd_soc_ops neo1973_hifi_ops = {
++	.hw_params = neo1973_hifi_hw_params,
++	.hw_free = neo1973_hifi_hw_free,
++};
++
++static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++	unsigned int pcmdiv = 0;
++	int ret = 0;
++	unsigned long iis_clkrate;
++
++	iis_clkrate = s3c24xx_i2s_get_clockrate();
++
++	if (params_rate(params) != 8000)
++		return -EINVAL;
++	if (params_channels(params) != 1)
++		return -EINVAL;
++
++	pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
++
++	/* todo: gg check mode (DSP_B) against CSR datasheet */
++	/* set codec DAI configuration */
++	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
++		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++	if (ret < 0)
++		return ret;
++
++	/* set the codec system clock for DAC and ADC */
++	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
++		SND_SOC_CLOCK_IN);
++	if (ret < 0)
++		return ret;
++
++	/* set codec PCM division for sample rate */
++	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
++	if (ret < 0)
++		return ret;
++
++	/* configue and enable PLL for 12.288MHz output */
++	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2,
++		iis_clkrate / 4, 12288000);
++	if (ret < 0)
++		return ret;
++
++	return 0;
++}
++
++static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++	/* disable the PLL */
++	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
++}
++
++static struct snd_soc_ops neo1973_voice_ops = {
++	.hw_params = neo1973_voice_hw_params,
++	.hw_free = neo1973_voice_hw_free,
++};
++
++static int neo1973_scenario = 0;
++
++static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	ucontrol->value.integer.value[0] = neo1973_scenario;
++	return 0;
++}
++
++static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
++{
++	switch(neo1973_scenario) {
++	case NEO_AUDIO_OFF:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_GSM_CALL_AUDIO_HANDSET:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     1);
++		break;
++	case NEO_GSM_CALL_AUDIO_HEADSET:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  1);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_GSM_CALL_AUDIO_BLUETOOTH:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_STEREO_TO_SPEAKERS:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_STEREO_TO_HEADPHONES:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_CAPTURE_HANDSET:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     1);
++		break;
++	case NEO_CAPTURE_HEADSET:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  1);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	case NEO_CAPTURE_BLUETOOTH:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++		break;
++	default:
++		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
++		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
++		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
++	}
++
++	snd_soc_dapm_sync_endpoints(codec);
++
++	return 0;
++}
++
++static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++
++	if (neo1973_scenario == ucontrol->value.integer.value[0])
++		return 0;
++
++	neo1973_scenario = ucontrol->value.integer.value[0];
++	set_scenario_endpoints(codec, neo1973_scenario);
++	return 1;
++}
++
++static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
++
++static void lm4857_write_regs(void)
++{
++	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
++		printk(KERN_ERR "lm4857: i2c write failed\n");
++}
++
++static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	int reg=kcontrol->private_value & 0xFF;
++	int shift = (kcontrol->private_value >> 8) & 0x0F;
++	int mask = (kcontrol->private_value >> 16) & 0xFF;
++
++	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
++	return 0;
++}
++
++static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	int reg = kcontrol->private_value & 0xFF;
++	int shift = (kcontrol->private_value >> 8) & 0x0F;
++	int mask = (kcontrol->private_value >> 16) & 0xFF;
++
++	if (((lm4857_regs[reg] >> shift ) & mask) ==
++		ucontrol->value.integer.value[0])
++		return 0;
++
++	lm4857_regs[reg] &= ~ (mask << shift);
++	lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
++	lm4857_write_regs();
++	return 1;
++}
++
++static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
++
++	if (value)
++		value -= 5;
++
++	ucontrol->value.integer.value[0] = value;
++	return 0;
++}
++
++static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
++	struct snd_ctl_elem_value *ucontrol)
++{
++	u8 value = ucontrol->value.integer.value[0];
++
++	if (value)
++		value += 5;
++
++	if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
++		return 0;
++
++	lm4857_regs[LM4857_CTRL] &= 0xF0;
++	lm4857_regs[LM4857_CTRL] |= value;
++	lm4857_write_regs();
++	return 1;
++}
++
++static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
++	SND_SOC_DAPM_LINE("Audio Out", NULL),
++	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
++	SND_SOC_DAPM_LINE("GSM Line In", NULL),
++	SND_SOC_DAPM_MIC("Headset Mic", NULL),
++	SND_SOC_DAPM_MIC("Call Mic", NULL),
++};
++
++
++/* example machine audio_mapnections */
++static const char* audio_map[][3] = {
++
++	/* Connections to the lm4857 amp */
++	{"Audio Out", NULL, "LOUT1"},
++	{"Audio Out", NULL, "ROUT1"},
++
++	/* Connections to the GSM Module */
++	{"GSM Line Out", NULL, "MONO1"},
++	{"GSM Line Out", NULL, "MONO2"},
++	{"RXP", NULL, "GSM Line In"},
++	{"RXN", NULL, "GSM Line In"},
++
++	/* Connections to Headset */
++	{"MIC1", NULL, "Mic Bias"},
++	{"Mic Bias", NULL, "Headset Mic"},
++
++	/* Call Mic */
++	{"MIC2", NULL, "Mic Bias"},
++	{"MIC2N", NULL, "Mic Bias"},
++	{"Mic Bias", NULL, "Call Mic"},
++
++	/* Connect the ALC pins */
++	{"ACIN", NULL, "ACOP"},
++
++	{NULL, NULL, NULL},
++};
++
++static const char *lm4857_mode[] = {
++	"Off",
++	"Call Speaker",
++	"Stereo Speakers",
++	"Stereo Speakers + Headphones",
++	"Headphones"
++};
++
++static const struct soc_enum lm4857_mode_enum[] = {
++	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
++};
++
++static const char *neo_scenarios[] = {
++	"Off",
++	"GSM Handset",
++	"GSM Headset",
++	"GSM Bluetooth",
++	"Speakers",
++	"Headphones",
++	"Capture Handset",
++	"Capture Headset",
++	"Capture Bluetooth"
++};
++
++static const struct soc_enum neo_scenario_enum[] = {
++	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios),
++};
++
++static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
++	SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
++		lm4857_get_mode, lm4857_set_mode),
++	SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
++		neo1973_get_scenario, neo1973_set_scenario),
++	SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
++		lm4857_get_reg, lm4857_set_reg),
++	SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
++		lm4857_get_reg, lm4857_set_reg),
++};
++
++/*
++ * This is an example machine initialisation for a wm8753 connected to a
++ * neo1973 II. It is missing logic to detect hp/mic insertions and logic
++ * to re-route the audio in such an event.
++ */
++static int neo1973_wm8753_init(struct snd_soc_codec *codec)
++{
++	int i, err;
++
++	/* set up NC codec pins */
++	snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
++	snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
++	snd_soc_dapm_set_endpoint(codec, "OUT3",  0);
++	snd_soc_dapm_set_endpoint(codec, "OUT4",  0);
++	snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
++	snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
++
++
++	/* set endpoints to default mode */
++	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
++
++	/* Add neo1973 specific widgets */
++	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
++		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
++
++	/* add neo1973 specific controls */
++	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
++		err = snd_ctl_add(codec->card,
++				snd_soc_cnew(&wm8753_neo1973_controls[i],
++				codec, NULL));
++		if (err < 0)
++			return err;
++	}
++
++	/* set up neo1973 specific audio path audio_mapnects */
++	for (i = 0; audio_map[i][0] != NULL; i++) {
++		snd_soc_dapm_connect_input(codec, audio_map[i][0],
++			audio_map[i][1], audio_map[i][2]);
++	}
++
++	snd_soc_dapm_sync_endpoints(codec);
++	return 0;
++}
++
++/*
++ * BT Codec DAI
++ */
++static struct snd_soc_cpu_dai bt_dai =
++{	.name = "Bluetooth",
++	.id = 0,
++	.type = SND_SOC_DAI_PCM,
++	.playback = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = SNDRV_PCM_RATE_8000,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = SNDRV_PCM_RATE_8000,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++};
++
++static struct snd_soc_dai_link neo1973_dai[] = {
++{ /* Hifi Playback - for similatious use with voice below */
++	.name = "WM8753",
++	.stream_name = "WM8753 HiFi",
++	.cpu_dai = &s3c24xx_i2s_dai,
++	.codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
++	.init = neo1973_wm8753_init,
++	.ops = &neo1973_hifi_ops,
++},
++{ /* Voice via BT */
++	.name = "Bluetooth",
++	.stream_name = "Voice",
++	.cpu_dai = &bt_dai,
++	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++	.ops = &neo1973_voice_ops,
++},
++};
++
++static struct snd_soc_machine neo1973 = {
++	.name = "neo1973",
++	.dai_link = neo1973_dai,
++	.num_links = ARRAY_SIZE(neo1973_dai),
++};
++
++static struct wm8753_setup_data neo1973_wm8753_setup = {
++	.i2c_address = 0x1a,
++};
++
++static struct snd_soc_device neo1973_snd_devdata = {
++	.machine = &neo1973,
++	.platform = &s3c24xx_soc_platform,
++	.codec_dev = &soc_codec_dev_wm8753,
++	.codec_data = &neo1973_wm8753_setup,
++};
++
++static struct i2c_client client_template;
++
++static unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++	int ret;
++
++	client_template.adapter = adap;
++	client_template.addr = addr;
++
++	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
++	if (i2c == NULL)
++		return -ENOMEM;
++
++	ret = i2c_attach_client(i2c);
++	if (ret < 0) {
++		printk(KERN_ERR "LM4857 failed to attach at addr %x\n", addr);
++		goto exit_err;
++	}
++
++	lm4857_write_regs();
++	return ret;
++
++exit_err:
++	kfree(i2c);
++	return ret;
++}
++
++static int lm4857_i2c_detach(struct i2c_client *client)
++{
++	i2c_detach_client(client);
++	kfree(client);
++	return 0;
++}
++
++static int lm4857_i2c_attach(struct i2c_adapter *adap)
++{
++	return i2c_probe(adap, &addr_data, lm4857_amp_probe);
++}
++
++/* corgi i2c codec control layer */
++static struct i2c_driver lm4857_i2c_driver = {
++	.driver = {
++		.name = "LM4857 I2C Amp",
++		.owner = THIS_MODULE,
++	},
++	.id =             I2C_DRIVERID_LM4857,
++	.attach_adapter = lm4857_i2c_attach,
++	.detach_client =  lm4857_i2c_detach,
++	.command =        NULL,
++};
++
++static struct i2c_client client_template = {
++	.name =   "LM4857",
++	.driver = &lm4857_i2c_driver,
++};
++
++static struct platform_device *neo1973_snd_device;
++
++static int __init neo1973_init(void)
++{
++	int ret;
++
++	neo1973_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!neo1973_snd_device)
++		return -ENOMEM;
++
++	platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata);
++	neo1973_snd_devdata.dev = &neo1973_snd_device->dev;
++	ret = platform_device_add(neo1973_snd_device);
++
++	if (ret)
++		platform_device_put(neo1973_snd_device);
++
++	ret = i2c_add_driver(&lm4857_i2c_driver);
++	if (ret != 0)
++		printk(KERN_ERR "can't add i2c driver");
++
++	return ret;
++}
++
++static void __exit neo1973_exit(void)
++{
++	platform_device_unregister(neo1973_snd_device);
++}
++
++module_init(neo1973_init);
++module_exit(neo1973_exit);
++
++/* Module information */
++MODULE_AUTHOR("Graeme Gregory, graeme.gregory at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
++MODULE_LICENSE("GPL");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/s3c24xx/s3c2443-ac97.c
+@@ -0,0 +1,401 @@
++/*
++ * s3c2443-ac97.c  --  ALSA Soc Audio Layer
++ *
++ * (c) 2007 Wolfson Microelectronics PLC.
++ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  Copyright (C) 2005, Sean Choi <sh428.choi at samsung.com>
++ *  All rights reserved.
++ *
++ *  This program is free software; you can redistribute it and/or modify
++ *  it under the terms of the GNU General Public License version 2 as
++ *  published by the Free Software Foundation.
++ *
++ *  Revision history
++ *	21st Mar 2007   Initial Version
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++#include <linux/interrupt.h>
++#include <linux/wait.h>
++#include <linux/delay.h>
++#include <linux/clk.h>
++
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++#include <asm/hardware.h>
++#include <asm/io.h>
++#include <asm/arch/regs-ac97.h>
++#include <asm/arch/regs-gpio.h>
++#include <asm/arch/regs-clock.h>
++#include <asm/arch/audio.h>
++#include <asm/dma.h>
++#include <asm/arch/dma.h>
++
++#include "s3c24xx-pcm.h"
++#include "s3c24xx-ac97.h"
++
++struct s3c24xx_ac97_info {
++	void __iomem	*regs;
++	struct clk	*ac97_clk;
++};
++static struct s3c24xx_ac97_info s3c24xx_ac97;
++
++DECLARE_COMPLETION(ac97_completion);
++static u32 codec_ready;
++static DECLARE_MUTEX(ac97_mutex);
++
++static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
++	unsigned short reg)
++{
++	u32 ac_glbctrl;
++	u32 ac_codec_cmd;
++	u32 stat, addr, data;
++
++	down(&ac97_mutex);
++
++	codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
++	ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++	ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg);
++	writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++
++	udelay(50);
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++
++	wait_for_completion(&ac97_completion);
++
++	stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT);
++	addr = (stat >> 16) & 0x7f;
++	data = (stat & 0xffff);
++
++	if (addr != reg)
++		printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
++				" rep addr = %02x\n", reg, addr);
++
++	up(&ac97_mutex);
++
++	return (unsigned short)data;
++}
++
++static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
++	unsigned short val)
++{
++	u32 ac_glbctrl;
++	u32 ac_codec_cmd;
++
++	down(&ac97_mutex);
++
++	codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
++	ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++	ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val);
++	writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++
++	udelay(50);
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++
++	wait_for_completion(&ac97_completion);
++
++	ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++	ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
++	writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
++
++	up(&ac97_mutex);
++
++}
++
++static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97)
++{
++	u32 ac_glbctrl;
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl = 0;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++}
++
++static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97)
++{
++	u32 ac_glbctrl;
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl = 0;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA |
++		S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++}
++
++static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id)
++{
++	int status;
++	u32 ac_glbctrl;
++
++	status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready;
++
++	if (status) {
++		ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++		ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE;
++		writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++		complete(&ac97_completion);
++	}
++	return IRQ_HANDLED;
++}
++
++struct snd_ac97_bus_ops soc_ac97_ops = {
++	.read	= s3c2443_ac97_read,
++	.write	= s3c2443_ac97_write,
++	.warm_reset	= s3c2443_ac97_warm_reset,
++	.reset	= s3c2443_ac97_cold_reset,
++};
++
++static struct s3c2410_dma_client s3c2443_dma_client_out = {
++	.name = "AC97 PCM Stereo out"
++};
++
++static struct s3c2410_dma_client s3c2443_dma_client_in = {
++	.name = "AC97 PCM Stereo in"
++};
++
++static struct s3c2410_dma_client s3c2443_dma_client_micin = {
++	.name = "AC97 Mic Mono in"
++};
++
++static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = {
++	.client		= &s3c2443_dma_client_out,
++	.channel	= DMACH_PCM_OUT,
++	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
++	.dma_size	= 4,
++};
++
++static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = {
++	.client		= &s3c2443_dma_client_in,
++	.channel	= DMACH_PCM_IN,
++	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
++	.dma_size	= 4,
++};
++
++static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
++	.client		= &s3c2443_dma_client_micin,
++	.channel	= DMACH_MIC_IN,
++	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_MIC_DATA,
++	.dma_size	= 4,
++};
++
++static int s3c2443_ac97_probe(struct platform_device *pdev)
++{
++	int ret;
++	u32 ac_glbctrl;
++
++	s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100);
++	if (s3c24xx_ac97.regs == NULL)
++		return -ENXIO;
++
++	s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97");
++	if (s3c24xx_ac97.ac97_clk == NULL) {
++		printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n");
++		iounmap(s3c24xx_ac97.regs);
++		return -ENODEV;
++	}
++	clk_enable(s3c24xx_ac97.ac97_clk);
++
++	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET);
++	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC);
++	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK);
++	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI);
++	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO);
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl = 0;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	msleep(1);
++
++	ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++
++	ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq,
++		IRQF_DISABLED, "AC97", NULL);
++	if (ret < 0) {
++		printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n");
++		clk_disable(s3c24xx_ac97.ac97_clk);
++		clk_put(s3c24xx_ac97.ac97_clk);
++		iounmap(s3c24xx_ac97.regs);
++	}
++	return ret;
++}
++
++static void s3c2443_ac97_remove(struct platform_device *pdev)
++{
++	free_irq(IRQ_S3C2443_AC97, NULL);
++	clk_disable(s3c24xx_ac97.ac97_clk);
++	clk_put(s3c24xx_ac97.ac97_clk);
++	iounmap(s3c24xx_ac97.regs);
++}
++
++static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
++				struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out;
++	else
++		cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in;
++
++	return 0;
++}
++
++static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	u32 ac_glbctrl;
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	switch(cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++	case SNDRV_PCM_TRIGGER_RESUME:
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++			ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
++		else
++			ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA;
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
++			ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
++		else
++			ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK;
++		break;
++	}
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++
++	return 0;
++}
++
++static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
++	struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
++		return -ENODEV;
++	else
++		cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in;
++
++	return 0;
++}
++
++static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
++	int cmd)
++{
++	u32 ac_glbctrl;
++
++	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++	switch(cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++	case SNDRV_PCM_TRIGGER_RESUME:
++	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
++		ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++	case SNDRV_PCM_TRIGGER_SUSPEND:
++	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
++		ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
++	}
++	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
++
++	return 0;
++}
++
++#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
++		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
++		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
++
++struct snd_soc_cpu_dai s3c2443_ac97_dai[] = {
++{
++	.name = "s3c2443-ac97",
++	.id = 0,
++	.type = SND_SOC_DAI_AC97,
++	.probe = s3c2443_ac97_probe,
++	.remove = s3c2443_ac97_remove,
++	.playback = {
++		.stream_name = "AC97 Playback",
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = s3c2443_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.capture = {
++		.stream_name = "AC97 Capture",
++		.channels_min = 2,
++		.channels_max = 2,
++		.rates = s3c2443_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = s3c2443_ac97_hw_params,
++		.trigger = s3c2443_ac97_trigger},
++},
++{
++	.name = "pxa2xx-ac97-mic",
++	.id = 1,
++	.type = SND_SOC_DAI_AC97,
++	.capture = {
++		.stream_name = "AC97 Mic Capture",
++		.channels_min = 1,
++		.channels_max = 1,
++		.rates = s3c2443_AC97_RATES,
++		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
++	.ops = {
++		.hw_params = s3c2443_ac97_hw_mic_params,
++		.trigger = s3c2443_ac97_mic_trigger,},
++},
++};
++
++EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
++EXPORT_SYMBOL_GPL(soc_ac97_ops);
++
++MODULE_AUTHOR("Graeme Gregory");
++MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
++MODULE_LICENSE("GPL");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/s3c24xx/s3c24xx-ac97.h
+@@ -0,0 +1,25 @@
++/*
++ * s3c24xx-ac97.c  --  ALSA Soc Audio Layer
++ *
++ * (c) 2007 Wolfson Microelectronics PLC.
++ * Author: Graeme Gregory
++ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    10th Nov 2006   Initial version.
++ */
++
++#ifndef S3C24XXAC97_H_
++#define S3C24XXAC97_H_
++
++#define AC_CMD_ADDR(x) (x << 16)
++#define AC_CMD_DATA(x) (x & 0xffff)
++
++extern struct snd_soc_cpu_dai s3c2443_ac97_dai[];
++
++#endif /*S3C24XXAC97_H_*/
+--- linux-2.6.22.1.orig/sound/soc/s3c24xx/s3c24xx-i2s.c
++++ linux-2.6.22.1/sound/soc/s3c24xx/s3c24xx-i2s.c
+@@ -344,11 +344,11 @@
+ 	DBG("Entered %s\n", __FUNCTION__);
+ 
+ 	switch (div_id) {
+-	case S3C24XX_DIV_MCLK:
++	case S3C24XX_DIV_BCLK:
+ 		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK;
+ 		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ 		break;
+-	case S3C24XX_DIV_BCLK:
++	case S3C24XX_DIV_MCLK:
+ 		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS);
+ 		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ 		break;
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/s3c24xx/smdk2443_wm9710.c
+@@ -0,0 +1,85 @@
++/*
++ * smdk2443_wm9710.c  --  SoC audio for smdk2443
++ *
++ * Copyright 2007 Wolfson Microelectronics PLC.
++ * Author: Graeme Gregory
++ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ *  This program is free software; you can redistribute  it and/or modify it
++ *  under  the terms of  the GNU General  Public License as published by the
++ *  Free Software Foundation;  either version 2 of the  License, or (at your
++ *  option) any later version.
++ *
++ *  Revision history
++ *    8th Mar 2007   Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include "../codecs/ac97.h"
++#include "s3c24xx-pcm.h"
++#include "s3c24xx-ac97.h"
++
++static struct snd_soc_machine smdk2443;
++
++static struct snd_soc_dai_link smdk2443_dai[] = {
++{
++	.name = "AC97",
++	.stream_name = "AC97 HiFi",
++	.cpu_dai = &s3c2443_ac97_dai[0],
++	.codec_dai = &ac97_dai,
++},
++};
++
++static struct snd_soc_machine smdk2443 = {
++	.name = "SMDK2443",
++	.dai_link = smdk2443_dai,
++	.num_links = ARRAY_SIZE(smdk2443_dai),
++};
++
++static struct snd_soc_device smdk2443_snd_ac97_devdata = {
++	.machine = &smdk2443,
++	.platform = &s3c24xx_soc_platform,
++	.codec_dev = &soc_codec_dev_ac97,
++};
++
++static struct platform_device *smdk2443_snd_ac97_device;
++
++static int __init smdk2443_init(void)
++{
++	int ret;
++
++	smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1);
++	if (!smdk2443_snd_ac97_device)
++		return -ENOMEM;
++
++	platform_set_drvdata(smdk2443_snd_ac97_device,
++				&smdk2443_snd_ac97_devdata);
++	smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev;
++	ret = platform_device_add(smdk2443_snd_ac97_device);
++
++	if (ret)
++		platform_device_put(smdk2443_snd_ac97_device);
++
++	return ret;
++}
++
++static void __exit smdk2443_exit(void)
++{
++	platform_device_unregister(smdk2443_snd_ac97_device);
++}
++
++module_init(smdk2443_init);
++module_exit(smdk2443_exit);
++
++/* Module information */
++MODULE_AUTHOR("Graeme Gregory, graeme.gregory at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443");
++MODULE_LICENSE("GPL");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/Kconfig
+@@ -0,0 +1,38 @@
++menu "SoC Audio support for SuperH"
++
++config SND_SOC_PCM_SH7760
++	tristate "SoC Audio support for Renesas SH7760"
++	depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG
++	help
++	  Enable this option for SH7760 AC97/I2S audio support.
++
++
++##
++## Audio unit modules
++##
++
++config SND_SOC_SH4_HAC
++	select AC97_BUS
++	select SND_SOC_AC97_BUS
++	select SND_AC97_CODEC
++	tristate
++
++config SND_SOC_SH4_SSI
++	tristate
++
++
++
++##
++## Boards
++##
++
++config SND_SH7760_AC97
++	tristate "SH7760 AC97 sound support"
++	depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760
++	select SND_SOC_SH4_HAC
++	select SND_SOC_AC97_CODEC
++	help
++	  This option enables generic sound support for the first
++	  AC97 unit of the SH7760.
++
++endmenu
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/Makefile
+@@ -0,0 +1,14 @@
++## DMA engines
++snd-soc-dma-sh7760-objs	:= dma-sh7760.o
++obj-$(CONFIG_SND_SOC_PCM_SH7760)	+= snd-soc-dma-sh7760.o
++
++## audio units found on some SH-4
++snd-soc-hac-objs	:= hac.o
++snd-soc-ssi-objs	:= ssi.o
++obj-$(CONFIG_SND_SOC_SH4_HAC)	+= snd-soc-hac.o
++obj-$(CONFIG_SND_SOC_SH4_SSI)	+= snd-soc-ssi.o
++
++## boards
++snd-soc-sh7760-ac97-objs	:= sh7760-ac97.o
++
++obj-$(CONFIG_SND_SH7760_AC97)	+= snd-soc-sh7760-ac97.o
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/dma-sh7760.c
+@@ -0,0 +1,354 @@
++/*
++ * SH7760 ("camelot") DMABRG audio DMA unit support
++ *
++ * Copyright (C) 2007 Manuel Lauss <mano at roarinelk.homelinux.net>
++ *  licensed under the terms outlined in the file COPYING at the root
++ *  of the linux kernel sources.
++ *
++ * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
++ * trigger an interrupt when one half of the programmed transfer size
++ * has been xmitted.
++ *
++ * FIXME: little-endian only for now
++ */
++
++#include <linux/module.h>
++#include <linux/init.h>
++#include <linux/platform_device.h>
++#include <linux/dma-mapping.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <asm/dmabrg.h>
++
++
++/* registers and bits */
++#define BRGATXSAR	0x00
++#define BRGARXDAR	0x04
++#define BRGATXTCR	0x08
++#define BRGARXTCR	0x0C
++#define BRGACR		0x10
++#define BRGATXTCNT	0x14
++#define BRGARXTCNT	0x18
++
++#define ACR_RAR		(1 << 18)
++#define ACR_RDS		(1 << 17)
++#define ACR_RDE		(1 << 16)
++#define ACR_TAR		(1 << 2)
++#define ACR_TDS		(1 << 1)
++#define ACR_TDE		(1 << 0)
++
++/* receiver/transmitter data alignment */
++#define ACR_RAM_NONE	(0 << 24)
++#define ACR_RAM_4BYTE	(1 << 24)
++#define ACR_RAM_2WORD	(2 << 24)
++#define ACR_TAM_NONE	(0 << 8)
++#define ACR_TAM_4BYTE	(1 << 8)
++#define ACR_TAM_2WORD	(2 << 8)
++
++
++struct camelot_pcm {
++	unsigned long mmio;  /* DMABRG audio channel control reg MMIO */
++	unsigned int txid;    /* ID of first DMABRG IRQ for this unit */
++
++	struct snd_pcm_substream *tx_ss;
++	unsigned long tx_period_size;
++	unsigned int  tx_period;
++
++	struct snd_pcm_substream *rx_ss;
++	unsigned long rx_period_size;
++	unsigned int  rx_period;
++
++} cam_pcm_data[2] = {
++	{
++		.mmio	=	0xFE3C0040,
++		.txid	=	DMABRGIRQ_A0TXF,
++	},
++	{
++		.mmio	=	0xFE3C0060,
++		.txid	=	DMABRGIRQ_A1TXF,
++	},
++};
++
++#define BRGREG(x)	(*(unsigned long *)(cam->mmio + (x)))
++
++/*
++ * set a minimum of 16kb per period, to avoid interrupt-"storm" and
++ * resulting skipping. In general, the bigger the minimum size, the
++ * better for overall system performance. (The SH7760 is a puny CPU
++ * with a slow SDRAM interface and poor internal bus bandwidth,
++ * *especially* when the LCDC is active).  The minimum for the DMAC
++ * is 8 bytes; 16kbytes are enough to get skip-free playback of a
++ * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain
++ * reasonable responsiveness in MPlayer.
++ */
++#define DMABRG_PERIOD_MIN		16 * 1024
++#define DMABRG_PERIOD_MAX		0x03fffffc
++#define DMABRG_PREALLOC_BUFFER		32 * 1024
++#define DMABRG_PREALLOC_BUFFER_MAX	32 * 1024
++
++/* support everything the SSI supports */
++#define DMABRG_RATES	\
++	SNDRV_PCM_RATE_8000_192000
++
++#define DMABRG_FMTS	\
++	(SNDRV_PCM_FMTBIT_S8      | SNDRV_PCM_FMTBIT_U8      |	\
++	 SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_U16_LE  |	\
++	 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE |	\
++	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
++	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
++
++static struct snd_pcm_hardware camelot_pcm_hardware = {
++	.info = (SNDRV_PCM_INFO_MMAP |
++		SNDRV_PCM_INFO_INTERLEAVED |
++		SNDRV_PCM_INFO_BLOCK_TRANSFER |
++		SNDRV_PCM_INFO_MMAP_VALID),
++	.formats =	DMABRG_FMTS,
++	.rates =	DMABRG_RATES,
++	.rate_min =		8000,
++	.rate_max =		192000,
++	.channels_min =		2,
++	.channels_max =		8,		/* max of the SSI */
++	.buffer_bytes_max =	DMABRG_PERIOD_MAX,
++	.period_bytes_min =	DMABRG_PERIOD_MIN,
++	.period_bytes_max =	DMABRG_PERIOD_MAX / 2,
++	.periods_min =		2,
++	.periods_max =		2,
++	.fifo_size =		128,
++};
++
++static void camelot_txdma(void *data)
++{
++	struct camelot_pcm *cam = data;
++	cam->tx_period ^= 1;
++	snd_pcm_period_elapsed(cam->tx_ss);
++}
++
++static void camelot_rxdma(void *data)
++{
++	struct camelot_pcm *cam = data;
++	cam->rx_period ^= 1;
++	snd_pcm_period_elapsed(cam->rx_ss);
++}
++
++static int camelot_pcm_open(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
++	int ret, dmairq;
++
++	snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware);
++
++	/* DMABRG buffer half/full events */
++	dmairq = (recv) ? cam->txid + 2 : cam->txid;
++	if (recv) {
++		cam->rx_ss = substream;
++		ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam);
++		if (unlikely(ret)) {
++			pr_debug("audio unit %d irqs already taken!\n",
++			     rtd->dai->cpu_dai->id);
++			return -EBUSY;
++		}
++		(void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam);
++	} else {
++		cam->tx_ss = substream;
++		ret = dmabrg_request_irq(dmairq, camelot_txdma, cam);
++		if (unlikely(ret)) {
++			pr_debug("audio unit %d irqs already taken!\n",
++			     rtd->dai->cpu_dai->id);
++			return -EBUSY;
++		}
++		(void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam);
++	}
++	return 0;
++}
++
++static int camelot_pcm_close(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
++	int dmairq;
++
++	dmairq = (recv) ? cam->txid + 2 : cam->txid;
++
++	if (recv)
++		cam->rx_ss = NULL;
++	else
++		cam->tx_ss = NULL;
++
++	dmabrg_free_irq(dmairq + 1);
++	dmabrg_free_irq(dmairq);
++
++	return 0;
++}
++
++static int camelot_hw_params(struct snd_pcm_substream *substream,
++			     struct snd_pcm_hw_params *hw_params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
++	int ret;
++
++	ret = snd_pcm_lib_malloc_pages(substream,
++				       params_buffer_bytes(hw_params));
++	if (ret < 0)
++		return ret;
++
++	if (recv) {
++		cam->rx_period_size = params_period_bytes(hw_params);
++		cam->rx_period = 0;
++	} else {
++		cam->tx_period_size = params_period_bytes(hw_params);
++		cam->tx_period = 0;
++	}
++	return 0;
++}
++
++static int camelot_hw_free(struct snd_pcm_substream *substream)
++{
++	return snd_pcm_lib_free_pages(substream);
++}
++
++static int camelot_prepare(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++
++	pr_debug("PCM data: addr 0x%08ulx len %d\n",
++		 (u32)runtime->dma_addr, runtime->dma_bytes);
++ 
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area;
++		BRGREG(BRGATXTCR) = runtime->dma_bytes;
++	} else {
++		BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area;
++		BRGREG(BRGARXTCR) = runtime->dma_bytes;
++	}
++
++	return 0;
++}
++
++static inline void dmabrg_play_dma_start(struct camelot_pcm *cam)
++{
++	unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
++	/* start DMABRG engine: XFER start, auto-addr-reload */
++	BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD;
++}
++
++static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam)
++{
++	unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
++	/* forcibly terminate data transmission */
++	BRGREG(BRGACR) = acr | ACR_TDS;
++}
++
++static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam)
++{
++	unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
++	/* start DMABRG engine: recv start, auto-reload */
++	BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD;
++}
++
++static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam)
++{
++	unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
++	/* forcibly terminate data receiver */
++	BRGREG(BRGACR) = acr | ACR_RDS;
++}
++
++static int camelot_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		if (recv)
++			dmabrg_rec_dma_start(cam);
++		else
++			dmabrg_play_dma_start(cam);
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		if (recv)
++			dmabrg_rec_dma_stop(cam);
++		else
++			dmabrg_play_dma_stop(cam);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream)
++{
++	struct snd_pcm_runtime *runtime = substream->runtime;
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
++	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
++	unsigned long pos;
++
++	/* cannot use the DMABRG pointer register: under load, by the
++	 * time ALSA comes around to read the register, it is already
++	 * far ahead (or worse, already done with the fragment) of the
++	 * position at the time the IRQ was triggered, which results in
++	 * fast-playback sound in my test application (ScummVM)
++	 */
++	if (recv)
++		pos = cam->rx_period ? cam->rx_period_size : 0;
++	else
++		pos = cam->tx_period ? cam->tx_period_size : 0;
++
++	return bytes_to_frames(runtime, pos);
++}
++
++static struct snd_pcm_ops camelot_pcm_ops = {
++	.open		= camelot_pcm_open,
++	.close		= camelot_pcm_close,
++	.ioctl		= snd_pcm_lib_ioctl,
++	.hw_params	= camelot_hw_params,
++	.hw_free	= camelot_hw_free,
++	.prepare	= camelot_prepare,
++	.trigger	= camelot_trigger,
++	.pointer	= camelot_pos,
++};
++
++static void camelot_pcm_free(struct snd_pcm *pcm)
++{
++	snd_pcm_lib_preallocate_free_for_all(pcm);
++}
++
++static int camelot_pcm_new(struct snd_card *card,
++			   struct snd_soc_codec_dai *dai,
++			   struct snd_pcm *pcm)
++{
++	/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
++	 * in MMAP mode (i.e. aplay -M)
++	 */
++	snd_pcm_lib_preallocate_pages_for_all(pcm,
++		SNDRV_DMA_TYPE_CONTINUOUS,
++		snd_dma_continuous_data(GFP_KERNEL),
++		DMABRG_PREALLOC_BUFFER,	DMABRG_PREALLOC_BUFFER_MAX);
++
++	return 0;
++}
++
++struct snd_soc_platform sh7760_soc_platform = {
++	.name		= "sh7760-pcm",
++	.pcm_ops 	= &camelot_pcm_ops,
++	.pcm_new	= camelot_pcm_new,
++	.pcm_free	= camelot_pcm_free,
++};
++EXPORT_SYMBOL_GPL(sh7760_soc_platform);
++
++MODULE_LICENSE("GPL");
++MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
++MODULE_AUTHOR("Manuel Lauss <mano at roarinelk.homelinux.net>");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/hac.c
+@@ -0,0 +1,322 @@
++/*
++ * Hitachi Audio Controller (AC97) support for SH7760/SH7780
++ *
++ * Copyright (c) 2007 Manuel Lauss <mano at roarinelk.homelinux.net>
++ *  licensed under the terms outlined in the file COPYING at the root
++ *  of the linux kernel sources.
++ *
++ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
++ * enable HAC output pins!
++ */
++
++/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
++ * the FIRST can be used since ASoC does not pass any information to the
++ * ac97_read/write() functions regarding WHICH unit to use.  You'll have
++ * to edit the code a bit to use the other AC97 unit.		--mlau
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++#include <linux/interrupt.h>
++#include <linux/wait.h>
++#include <linux/delay.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/ac97_codec.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++
++/* regs and bits */
++#define HACCR		0x08
++#define HACCSAR		0x20
++#define HACCSDR		0x24
++#define HACPCML		0x28
++#define HACPCMR		0x2C
++#define HACTIER		0x50
++#define	HACTSR		0x54
++#define HACRIER		0x58
++#define HACRSR		0x5C
++#define HACACR		0x60
++
++#define CR_CR		(1 << 15)	/* "codec-ready" indicator */
++#define CR_CDRT		(1 << 11)	/* cold reset */
++#define CR_WMRT		(1 << 10)	/* warm reset */
++#define CR_B9		(1 << 9)	/* the mysterious "bit 9" */
++#define CR_ST		(1 << 5)	/* AC97 link start bit */
++
++#define CSAR_RD		(1 << 19)	/* AC97 data read bit */
++#define CSAR_WR		(0)
++
++#define TSR_CMDAMT	(1 << 31)
++#define TSR_CMDDMT	(1 << 30)
++
++#define RSR_STARY	(1 << 22)
++#define RSR_STDRY	(1 << 21)
++
++#define ACR_DMARX16	(1 << 30)
++#define ACR_DMATX16	(1 << 29)
++#define ACR_TX12ATOM	(1 << 26)
++#define ACR_DMARX20	((1 << 24) | (1 << 22))
++#define ACR_DMATX20	((1 << 23) | (1 << 21))
++
++#define CSDR_SHIFT	4
++#define CSDR_MASK	(0xffff << CSDR_SHIFT)
++#define CSAR_SHIFT	12
++#define CSAR_MASK	(0x7f << CSAR_SHIFT)
++
++#define AC97_WRITE_RETRY	1
++#define AC97_READ_RETRY		5
++
++/* manual-suggested AC97 codec access timeouts (us) */
++#define TMO_E1	500	/* 21 < E1 < 1000 */
++#define TMO_E2	13	/* 13 < E2 */
++#define TMO_E3	21	/* 21 < E3 */
++#define TMO_E4	500	/* 21 < E4 < 1000 */
++
++struct hac_priv {
++	unsigned long mmio;	/* HAC base address */
++} hac_cpu_data[] = {
++#if defined(CONFIG_CPU_SUBTYPE_SH7760)
++	{
++		.mmio	= 0xFE240000,
++	},
++	{
++		.mmio	= 0xFE250000,
++	},
++#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
++	{
++		.mmio	= 0xFFE40000,
++	},
++#else
++#error "Unsupported SuperH SoC"
++#endif
++};
++
++#define HACREG(reg)	(*(unsigned long *)(hac->mmio + (reg)))
++
++/*
++ * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906)
++ */
++static int hac_get_codec_data(struct hac_priv *hac, unsigned short r,
++			      unsigned short *v)
++{
++	unsigned int to1, to2, i;
++	unsigned short adr;
++
++	for (i = 0; i < AC97_READ_RETRY; ++i) {
++		*v = 0;
++		/* wait for HAC to receive something from the codec */
++		for (to1 = TMO_E4;
++		     to1 && !(HACREG(HACRSR) & RSR_STARY);
++		     --to1)
++			udelay(1);
++		for (to2 = TMO_E4; 
++		     to2 && !(HACREG(HACRSR) & RSR_STDRY);
++		     --to2)
++			udelay(1);
++
++		if (!to1 && !to2)
++			return 0;	/* codec comm is down */
++
++		adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT);
++		*v  = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT);
++
++		HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
++
++		if (r == adr)
++			break;
++
++		/* manual says: wait at least 21 usec before retrying */
++		udelay(21);
++	}
++	HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
++	return (i < AC97_READ_RETRY);
++}
++
++static unsigned short hac_read_codec_aux(struct hac_priv *hac,
++					 unsigned short reg)
++{
++	unsigned short val;
++	unsigned int i, to;
++
++	for (i = 0; i < AC97_READ_RETRY; i++) {
++		/* send_read_request */
++		local_irq_disable();
++		HACREG(HACTSR) &= ~(TSR_CMDAMT);
++		HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD;
++		local_irq_enable();
++
++		for (to = TMO_E3;
++		     to && !(HACREG(HACTSR) & TSR_CMDAMT);
++		     --to)
++			udelay(1);
++
++		HACREG(HACTSR) &= ~TSR_CMDAMT;
++		val = 0;
++		if (hac_get_codec_data(hac, reg, &val) != 0)
++			break;
++	}
++
++	if (i == AC97_READ_RETRY)
++		return ~0;
++
++	return val;
++}
++
++static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
++			   unsigned short val)
++{
++	int unit_id = 0 /* ac97->private_data */;
++	struct hac_priv *hac = &hac_cpu_data[unit_id];
++	unsigned int i, to;
++	/* write_codec_aux */
++	for (i = 0; i < AC97_WRITE_RETRY; i++) {
++		/* send_write_request */
++		local_irq_disable();
++		HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT);
++		HACREG(HACCSDR) = (val << CSDR_SHIFT);
++		HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD);
++		local_irq_enable();
++
++		/* poll-wait for CMDAMT and CMDDMT */
++		for (to = TMO_E1;
++		     to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT));
++		     --to)
++			udelay(1);
++
++		HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT);
++		if (to)
++			break;
++		/* timeout, try again */
++	}
++}
++
++static unsigned short hac_ac97_read(struct snd_ac97 *ac97,
++				    unsigned short reg)
++{
++	int unit_id = 0 /* ac97->private_data */;
++	struct hac_priv *hac = &hac_cpu_data[unit_id];
++	return hac_read_codec_aux(hac, reg);
++}
++
++static void hac_ac97_warmrst(struct snd_ac97 *ac97)
++{
++	int unit_id = 0 /* ac97->private_data */;
++	struct hac_priv *hac = &hac_cpu_data[unit_id];
++	unsigned int tmo;
++
++	HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9;
++	msleep(10);
++	HACREG(HACCR) = CR_ST | CR_B9;
++	for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--)
++		udelay(1);
++
++	if (!tmo)
++		printk(KERN_INFO "hac: reset: AC97 link down!\n");
++	/* settings this bit lets us have a conversation with codec */
++	HACREG(HACACR) |= ACR_TX12ATOM;
++}
++
++static void hac_ac97_coldrst(struct snd_ac97 *ac97)
++{
++	int unit_id = 0 /* ac97->private_data */;
++	struct hac_priv *hac;
++	hac = &hac_cpu_data[unit_id];
++
++	HACREG(HACCR) = 0;
++	HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9;
++	msleep(10);
++	hac_ac97_warmrst(ac97);
++}
++
++struct snd_ac97_bus_ops soc_ac97_ops = {
++	.read	= hac_ac97_read,
++	.write	= hac_ac97_write,
++	.reset	= hac_ac97_coldrst,
++	.warm_reset = hac_ac97_warmrst,
++};
++EXPORT_SYMBOL_GPL(soc_ac97_ops);
++
++static int hac_hw_params(struct snd_pcm_substream *substream,
++			 struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
++	int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
++
++	switch (params->msbits) {
++	case 16:
++		HACREG(HACACR) |= d ?  ACR_DMARX16 :  ACR_DMATX16;
++		HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20;
++		break;
++	case 20:
++		HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16;
++		HACREG(HACACR) |= d ?  ACR_DMARX20 :  ACR_DMATX20;
++		break;
++	default:
++		pr_debug("hac: invalid depth %d bit\n", params->msbits);
++		return -EINVAL;
++		break;
++	}
++
++	return 0;
++}
++
++#define AC97_RATES	\
++	SNDRV_PCM_RATE_8000_192000
++
++#define AC97_FMTS	\
++	SNDRV_PCM_FMTBIT_S16_LE
++
++struct snd_soc_cpu_dai sh4_hac_dai[] = {
++{
++	.name			= "HAC0",
++	.id			= 0,
++	.type			= SND_SOC_DAI_AC97,
++	.playback = {
++		.rates		= AC97_RATES,
++		.formats	= AC97_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 2,
++	},
++	.capture = {
++		.rates		= AC97_RATES,
++		.formats	= AC97_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 2,
++	},
++	.ops = {
++		.hw_params	= hac_hw_params,
++	},
++},
++#ifdef CONFIG_CPU_SUBTYPE_SH7760
++{
++	.name			= "HAC1",
++	.id			= 1,
++	.type			= SND_SOC_DAI_AC97,
++	.playback = {
++		.rates		= AC97_RATES,
++		.formats	= AC97_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 2,
++	},
++	.capture = {
++		.rates		= AC97_RATES,
++		.formats	= AC97_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 2,
++	},
++	.ops = {
++		.hw_params	= hac_hw_params,
++	},
++
++},
++#endif
++};
++EXPORT_SYMBOL_GPL(sh4_hac_dai);
++
++MODULE_LICENSE("GPL");
++MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
++MODULE_AUTHOR("Manuel Lauss <mano at roarinelk.homelinux.net>");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/sh7760-ac97.c
+@@ -0,0 +1,92 @@
++/*
++ * Generic AC97 sound support for SH7760
++ *
++ * (c) 2007 Manuel Lauss
++ *
++ * Licensed under the GPLv2.
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <asm/io.h>
++
++#include "../codecs/ac97.h"
++
++#define IPSEL 0xFE400034
++
++/* platform specific structs can be declared here */
++extern struct snd_soc_cpu_dai sh4_hac_dai[2];
++extern struct snd_soc_platform sh7760_soc_platform;
++
++static int machine_init(struct snd_soc_codec *codec)
++{
++	snd_soc_dapm_sync_endpoints(codec);
++	return 0;
++}
++
++static struct snd_soc_dai_link sh7760_ac97_dai = {
++	.name = "AC97",
++	.stream_name = "AC97 HiFi",
++	.cpu_dai = &sh4_hac_dai[0],	/* HAC0 */
++	.codec_dai = &ac97_dai,
++	.init = machine_init,
++	.ops = NULL,
++};
++
++static struct snd_soc_machine sh7760_ac97_soc_machine  = {
++	.name = "SH7760 AC97",
++	.dai_link = &sh7760_ac97_dai,
++	.num_links = 1,
++};
++
++static struct snd_soc_device sh7760_ac97_snd_devdata = {
++	.machine = &sh7760_ac97_soc_machine,
++	.platform = &sh7760_soc_platform,
++	.codec_dev = &soc_codec_dev_ac97,
++};
++
++static struct platform_device *sh7760_ac97_snd_device;
++
++static int __init sh7760_ac97_init(void)
++{
++	int ret;
++	unsigned short ipsel;
++
++	/* enable both AC97 controllers in pinmux reg */
++	ipsel = ctrl_inw(IPSEL);
++	ctrl_outw(ipsel | (3 << 10), IPSEL);
++
++	ret = -ENOMEM;
++	sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1);
++	if (!sh7760_ac97_snd_device)
++		goto out;
++
++	platform_set_drvdata(sh7760_ac97_snd_device,
++			     &sh7760_ac97_snd_devdata);
++	sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev;
++	ret = platform_device_add(sh7760_ac97_snd_device);
++
++	if (ret)
++		platform_device_put(sh7760_ac97_snd_device);
++
++out:
++	return ret;
++}
++
++static void __exit sh7760_ac97_exit(void)
++{
++	platform_device_unregister(sh7760_ac97_snd_device);
++}
++
++module_init(sh7760_ac97_init);
++module_exit(sh7760_ac97_exit);
++
++MODULE_LICENSE("GPL");
++MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
++MODULE_AUTHOR("Manuel Lauss <mano at roarinelk.homelinux.net>");
+--- /dev/null
++++ linux-2.6.22.1/sound/soc/sh/ssi.c
+@@ -0,0 +1,400 @@
++/*
++ * Serial Sound Interface (I2S) support for SH7760/SH7780
++ *
++ * Copyright (c) 2007 Manuel Lauss <mano at roarinelk.homelinux.net>
++ *
++ *  licensed under the terms outlined in the file COPYING at the root
++ *  of the linux kernel sources.
++ *
++ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
++ * enable SSI output pins!
++ */
++
++/*
++ * LIMITATIONS:
++ *	The SSI unit has only one physical data line, so full duplex is
++ *	impossible.  This can be remedied  on the  SH7760 by  using the
++ *	other SSI unit for recording; however the SH7780 has only 1 SSI
++ *	unit, and its pins are shared with the AC97 unit,  among others.
++ *
++ * FEATURES:
++ *	The SSI features "compressed mode": in this mode it continuously
++ *	streams PCM data over the I2S lines and uses LRCK as a handshake
++ *	signal.  Can be used to send compressed data (AC3/DTS) to a DSP.
++ *	The number of bits sent over the wire in a frame can be adjusted
++ *	and can be independent from the actual sample bit depth. This is
++ *	useful to support TDM mode codecs like the AD1939 which have a
++ *	fixed TDM slot size, regardless of sample resolution.
++ */
++
++#include <linux/init.h>
++#include <linux/module.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/initval.h>
++#include <sound/soc.h>
++#include <asm/io.h>
++
++#define SSICR	0x00
++#define SSISR	0x04
++
++#define CR_DMAEN	(1 << 28)
++#define CR_CHNL_SHIFT	22
++#define CR_CHNL_MASK	(3 << CR_CHNL_SHIFT)
++#define CR_DWL_SHIFT	19
++#define CR_DWL_MASK	(7 << CR_DWL_SHIFT)
++#define CR_SWL_SHIFT	16
++#define CR_SWL_MASK	(7 << CR_SWL_SHIFT)
++#define CR_SCK_MASTER	(1 << 15)	/* bitclock master bit */
++#define CR_SWS_MASTER	(1 << 14)	/* wordselect master bit */
++#define CR_SCKP		(1 << 13)	/* I2Sclock polarity */
++#define CR_SWSP		(1 << 12)	/* LRCK polarity */
++#define CR_SPDP		(1 << 11)
++#define CR_SDTA		(1 << 10)	/* i2s alignment (msb/lsb) */
++#define CR_PDTA		(1 << 9)	/* fifo data alignment */
++#define CR_DEL		(1 << 8)	/* delay data by 1 i2sclk */
++#define CR_BREN		(1 << 7)	/* clock gating in burst mode */
++#define CR_CKDIV_SHIFT	4
++#define CR_CKDIV_MASK	(7 << CR_CKDIV_SHIFT)	/* bitclock divider */
++#define CR_MUTE		(1 << 3)	/* SSI mute */
++#define CR_CPEN		(1 << 2)	/* compressed mode */
++#define CR_TRMD		(1 << 1)	/* transmit/receive select */
++#define CR_EN		(1 << 0)	/* enable SSI */
++
++#define SSIREG(reg)	(*(unsigned long *)(ssi->mmio + (reg)))
++
++struct ssi_priv {
++	unsigned long mmio;
++	unsigned long sysclk;
++	int inuse;
++} ssi_cpu_data[] = {
++#if defined(CONFIG_CPU_SUBTYPE_SH7760)
++	{
++		.mmio	= 0xFE680000,
++	},
++	{
++		.mmio	= 0xFE690000,
++	},
++#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
++	{
++		.mmio	= 0xFFE70000,
++	},
++#else
++#error "Unsupported SuperH SoC"
++#endif
++};
++
++/*
++ * track usage of the SSI; it is simplex-only so prevent attempts of
++ * concurrent playback + capture. FIXME: any locking required?
++ */
++static int ssi_startup(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
++	if (ssi->inuse) {
++		pr_debug("ssi: already in use!\n");
++		return -EBUSY;
++	} else
++		ssi->inuse = 1;
++	return 0;
++}
++
++static void ssi_shutdown(struct snd_pcm_substream *substream)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
++
++	ssi->inuse = 0;
++}
++
++static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
++
++	switch (cmd) {
++	case SNDRV_PCM_TRIGGER_START:
++		SSIREG(SSICR) |= CR_DMAEN | CR_EN;
++		break;
++	case SNDRV_PCM_TRIGGER_STOP:
++		SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN);
++		break;
++	default:
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int ssi_hw_params(struct snd_pcm_substream *substream,
++			 struct snd_pcm_hw_params *params)
++{
++	struct snd_soc_pcm_runtime *rtd = substream->private_data;
++	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
++	unsigned long ssicr = SSIREG(SSICR);
++	unsigned int bits, channels, swl, recv, i;
++
++	channels = params_channels(params);
++	bits = params->msbits;
++	recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
++
++	pr_debug("ssi_hw_params() enter\nssicr was    %08lx\n", ssicr);
++	pr_debug("bits: %d channels: %d\n", bits, channels);
++
++	ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
++		   CR_SWL_MASK);
++
++	/* direction (send/receive) */
++	if (!recv)
++		ssicr |= CR_TRMD;	/* transmit */
++
++	/* channels */
++	if ((channels < 2) || (channels > 8) || (channels & 1)) {
++		pr_debug("ssi: invalid number of channels\n");
++		return -EINVAL;
++	}
++	ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT;
++
++	/* DATA WORD LENGTH (DWL): databits in audio sample */
++	i = 0;
++	switch (bits) {
++	case 32: ++i;
++	case 24: ++i;
++	case 22: ++i;
++	case 20: ++i;
++	case 18: ++i;
++	case 16: ++i;
++		 ssicr |= i << CR_DWL_SHIFT;
++	case 8:	 break;
++	default:
++		pr_debug("ssi: invalid sample width\n");
++		return -EINVAL;
++	}
++
++	/*
++	 * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S
++	 * wires. This is usually bits_per_sample x channels/2;  i.e. in
++	 * Stereo mode  the SWL equals DWL.  SWL can  be bigger than the
++	 * product of (channels_per_slot x samplebits), e.g.  for codecs
++	 * like the AD1939 which  only accept 32bit wide TDM slots.  For
++	 * "standard" I2S operation we set SWL = chans / 2 * DWL here.
++	 * Waiting for ASoC to get TDM support ;-)
++	 */
++	if ((bits > 16) && (bits <= 24)) {
++		bits = 24;	/* these are padded by the SSI */
++		/*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */
++	}
++	i = 0;
++	swl = (bits * channels) / 2;
++	switch (swl) {
++	case 256: ++i;
++	case 128: ++i;
++	case 64:  ++i;
++	case 48:  ++i;
++	case 32:  ++i;
++	case 16:  ++i;
++		  ssicr |= i << CR_SWL_SHIFT;
++	case 8:   break;
++	default:
++		pr_debug("ssi: invalid system word length computed\n");
++		return -EINVAL;
++	}
++
++	SSIREG(SSICR) = ssicr;
++
++	pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr);
++	return 0;
++}
++
++static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
++			  unsigned int freq, int dir)
++{
++	struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id];
++
++	ssi->sysclk = freq;
++
++	return 0;
++}
++
++/*
++ * This divider is used to generate the SSI_SCK (I2S bitclock) from the
++ * clock at the HAC_BIT_CLK ("oversampling clock") pin.
++ */
++static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
++{
++	struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
++	unsigned long ssicr;
++	int i;
++
++	i = 0;
++	ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK;
++	switch (div) {
++	case 16: ++i;
++	case 8:  ++i;
++	case 4:  ++i;
++	case 2:  ++i;
++		 SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT);
++	case 1:  break;
++	default:
++		pr_debug("ssi: invalid sck divider %d\n", div);
++		return -EINVAL;
++	}
++
++	return 0;
++}
++
++static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
++{
++	struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
++	unsigned long ssicr = SSIREG(SSICR);
++
++	pr_debug("ssi_set_fmt()\nssicr was    0x%08lx\n", ssicr);
++
++	ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP |
++		   CR_SWS_MASTER | CR_SCK_MASTER);
++
++	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++	case SND_SOC_DAIFMT_I2S:
++		break;
++	case SND_SOC_DAIFMT_RIGHT_J:
++		ssicr |= CR_DEL | CR_PDTA;
++		break;
++	case SND_SOC_DAIFMT_LEFT_J:
++		ssicr |= CR_DEL;
++		break;
++	default:
++		pr_debug("ssi: unsupported format\n");
++		return -EINVAL;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
++	case SND_SOC_DAIFMT_CONT:
++		break;
++	case SND_SOC_DAIFMT_GATED:
++		ssicr |= CR_BREN;
++		break;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
++	case SND_SOC_DAIFMT_NB_NF:
++		ssicr |= CR_SCKP;	/* sample data at low clkedge */
++		break;
++	case SND_SOC_DAIFMT_NB_IF:
++		ssicr |= CR_SCKP | CR_SWSP;
++		break;
++	case SND_SOC_DAIFMT_IB_NF:
++		break;
++	case SND_SOC_DAIFMT_IB_IF:
++		ssicr |= CR_SWSP;	/* word select starts low */
++		break;
++	default:
++		pr_debug("ssi: invalid inversion\n");
++		return -EINVAL;
++	}
++
++	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
++	case SND_SOC_DAIFMT_CBM_CFM:
++		break;
++	case SND_SOC_DAIFMT_CBS_CFM:
++		ssicr |= CR_SCK_MASTER;
++		break;
++	case SND_SOC_DAIFMT_CBM_CFS:
++		ssicr |= CR_SWS_MASTER;
++		break;
++	case SND_SOC_DAIFMT_CBS_CFS:
++		ssicr |= CR_SWS_MASTER | CR_SCK_MASTER;
++		break;
++	default:
++		pr_debug("ssi: invalid master/slave configuration\n");
++		return -EINVAL;
++	}
++
++	SSIREG(SSICR) = ssicr;
++	pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr);
++
++	return 0;
++}
++
++/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in
++ * Master mode,  so really this is board specific;  the SSI can do any
++ * rate with the right bitclk and divider settings.
++ */
++#define SSI_RATES	\
++	SNDRV_PCM_RATE_8000_192000
++
++/* the SSI can do 8-32 bit samples, with 8 possible channels */
++#define SSI_FMTS	\
++	(SNDRV_PCM_FMTBIT_S8      | SNDRV_PCM_FMTBIT_U8      |	\
++	 SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_U16_LE  |	\
++	 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE |	\
++	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
++	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
++
++struct snd_soc_cpu_dai sh4_ssi_dai[] = {
++{
++	.name			= "SSI0",
++	.id			= 0,
++	.type			= SND_SOC_DAI_I2S,
++	.playback = {
++		.rates		= SSI_RATES,
++		.formats	= SSI_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 8,
++	},
++	.capture = {
++		.rates		= SSI_RATES,
++		.formats	= SSI_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 8,
++	},
++	.ops = {
++		.startup	= ssi_startup,
++		.shutdown	= ssi_shutdown,
++		.trigger	= ssi_trigger,
++		.hw_params	= ssi_hw_params,
++	},
++	.dai_ops = {
++		.set_sysclk	= ssi_set_sysclk,
++		.set_clkdiv	= ssi_set_clkdiv,
++		.set_fmt	= ssi_set_fmt,
++	},
++},
++#ifdef CONFIG_CPU_SUBTYPE_SH7760
++{
++	.name			= "SSI1",
++	.id			= 1,
++	.type			= SND_SOC_DAI_I2S,
++	.playback = {
++		.rates		= SSI_RATES,
++		.formats	= SSI_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 8,
++	},
++	.capture = {
++		.rates		= SSI_RATES,
++		.formats	= SSI_FMTS,
++		.channels_min	= 2,
++		.channels_max	= 8,
++	},
++	.ops = {
++		.startup	= ssi_startup,
++		.shutdown	= ssi_shutdown,
++		.trigger	= ssi_trigger,
++		.hw_params	= ssi_hw_params,
++	},
++	.dai_ops = {
++		.set_sysclk	= ssi_set_sysclk,
++		.set_clkdiv	= ssi_set_clkdiv,
++		.set_fmt	= ssi_set_fmt,
++	},
++},
++#endif
++};
++EXPORT_SYMBOL_GPL(sh4_ssi_dai);
++
++MODULE_LICENSE("GPL");
++MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
++MODULE_AUTHOR("Manuel Lauss <mano at roarinelk.homelinux.net>");
+--- linux-2.6.22.1.orig/sound/usb/usbaudio.c
++++ linux-2.6.22.1/sound/usb/usbaudio.c
+@@ -2350,7 +2350,9 @@
+ 			return 1;
+ 		break;
+ 	case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+-		return 1;
++		if (device_setup[chip->index] == 0x00 ||
++		    fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3)
++			return 1;
+ 	}
+ 	return 0;
+ }
+@@ -2530,7 +2532,18 @@
+ 		 *        but we give normal PCM format to get the existing
+ 		 *        apps working...
+ 		 */
+-		pcm_format = SNDRV_PCM_FORMAT_S16_LE;
++		switch (chip->usb_id) {
++
++		case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
++			if (device_setup[chip->index] == 0x00 && 
++			    fp->altsetting == 6)
++				pcm_format = SNDRV_PCM_FORMAT_S16_BE;
++			else
++				pcm_format = SNDRV_PCM_FORMAT_S16_LE;
++			break;
++		default:
++			pcm_format = SNDRV_PCM_FORMAT_S16_LE;
++		}
+ 	} else {
+ 		pcm_format = parse_audio_format_i_type(chip, fp, format, fmt);
+ 		if (pcm_format < 0)
+@@ -3251,6 +3264,11 @@
+ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+ 					 int iface, int altno)
+ {
++	/* Reset ALL ifaces to 0 altsetting.
++	 * Call it for every possible altsetting of every interface.
++	 */
++	usb_set_interface(chip->dev, iface, 0);
++
+ 	if (device_setup[chip->index] & AUDIOPHILE_SET) {
+ 		if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
+ 		    && altno != 6)
+--- linux-2.6.22.1.orig/sound/usb/usbquirks.h
++++ linux-2.6.22.1/sound/usb/usbquirks.h
+@@ -57,6 +57,24 @@
+ 		       USB_DEVICE_ID_MATCH_INT_CLASS |
+ 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ 	.idVendor = 0x046d,
++	.idProduct = 0x08ae,
++	.bInterfaceClass = USB_CLASS_AUDIO,
++	.bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
++},
++{
++	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
++		       USB_DEVICE_ID_MATCH_INT_CLASS |
++		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
++	.idVendor = 0x046d,
++	.idProduct = 0x08c6,
++	.bInterfaceClass = USB_CLASS_AUDIO,
++	.bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
++},
++{
++	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
++		       USB_DEVICE_ID_MATCH_INT_CLASS |
++		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
++	.idVendor = 0x046d,
+ 	.idProduct = 0x08f0,
+ 	.bInterfaceClass = USB_CLASS_AUDIO,
+ 	.bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+@@ -1051,7 +1069,15 @@
+ 		.type = QUIRK_MIDI_STANDARD_INTERFACE
+ 	}
+ },
+-	/* TODO: add Roland EXR support */
++{
++	USB_DEVICE(0x0582, 0x0060),
++	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
++		.vendor_name = "Roland",
++		.product_name = "EXR Series",
++		.ifnum = 0,
++		.type = QUIRK_MIDI_STANDARD_INTERFACE
++	}
++},
+ {
+ 	/* has ID 0x0067 when not in "Advanced Driver" mode */
+ 	USB_DEVICE(0x0582, 0x0065),
+@@ -1094,6 +1120,19 @@
+ 		}
+ 	}
+ },
++{
++	USB_DEVICE(0x582, 0x00a6),
++	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
++		.vendor_name = "Roland",
++		.product_name = "Juno-G",
++		.ifnum = 0,
++		.type = QUIRK_MIDI_FIXED_ENDPOINT,
++		.data = & (const struct snd_usb_midi_endpoint_info) {
++			.out_cables = 0x0001,
++			.in_cables  = 0x0001
++		}
++	}
++},
+ {	/*
+ 	 * This quirk is for the "Advanced" modes of the Edirol UA-25.
+ 	 * If the switch is not in an advanced setting, the UA-25 has
+@@ -1230,6 +1269,37 @@
+ 	}
+ },
+ 	/* TODO: add Edirol MD-P1 support */
++{
++	/* Roland SH-201 */
++	USB_DEVICE(0x0582, 0x00ad),
++	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
++		.vendor_name = "Roland",
++		.product_name = "SH-201",
++		.ifnum = QUIRK_ANY_INTERFACE,
++		.type = QUIRK_COMPOSITE,
++		.data = (const struct snd_usb_audio_quirk[]) {
++			{
++				.ifnum = 0,
++				.type = QUIRK_AUDIO_STANDARD_INTERFACE
++			},
++			{
++				.ifnum = 1,
++				.type = QUIRK_AUDIO_STANDARD_INTERFACE
++			},
++			{
++				.ifnum = 2,
++				.type = QUIRK_MIDI_FIXED_ENDPOINT,
++				.data = & (const struct snd_usb_midi_endpoint_info) {
++					.out_cables = 0x0001,
++					.in_cables  = 0x0001
++				}
++			},
++			{
++				.ifnum = -1
++			}
++		}
++	}
++},
+ 
+ /* Guillemot devices */
+ {
+--- linux-2.6.22.1.orig/sound/usb/usx2y/usbusx2yaudio.c
++++ linux-2.6.22.1/sound/usb/usx2y/usbusx2yaudio.c
+@@ -935,10 +935,9 @@
+  */
+ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
+ {
+-	if (NULL != usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]) {
+-		kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
+-		usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+-	}
++	kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
++	usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
++
+ 	kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
+ 	usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+ }

Deleted: branches/src/target/kernel/2.6.22.x/patches/asoc-asm_hardware_h.patch
===================================================================
--- branches/src/target/kernel/2.6.22.x/patches/asoc-asm_hardware_h.patch	2007-07-26 14:40:11 UTC (rev 2397)
+++ branches/src/target/kernel/2.6.22.x/patches/asoc-asm_hardware_h.patch	2007-07-26 15:24:28 UTC (rev 2398)
@@ -1,17 +0,0 @@
-Don't include <asm/arch/hardware.h>, but <asm/hardware.h>
-
-Signed-off-by: Harald Welte <laforge at openmko.org>
-
-Index: linux-2.6.20/sound/soc/s3c24xx/neo1973_wm8753.c
-===================================================================
---- linux-2.6.20.orig/sound/soc/s3c24xx/neo1973_wm8753.c	2007-02-15 16:27:53.000000000 +0100
-+++ linux-2.6.20/sound/soc/s3c24xx/neo1973_wm8753.c	2007-02-15 16:28:02.000000000 +0100
-@@ -33,7 +33,7 @@
- #include <asm/arch/regs-iis.h>
- #include <asm/arch/regs-clock.h>
- #include <asm/arch/regs-gpio.h>
--#include <asm/arch/hardware.h>
-+#include <asm/hardware.h>
- #include <asm/arch/audio.h>
- #include <asm/io.h>
- #include <asm/arch/spi-gpio.h>

Added: branches/src/target/kernel/2.6.22.x/patches/asoc-kconfig-fix.patch
===================================================================
--- branches/src/target/kernel/2.6.22.x/patches/asoc-kconfig-fix.patch	2007-07-26 14:40:11 UTC (rev 2397)
+++ branches/src/target/kernel/2.6.22.x/patches/asoc-kconfig-fix.patch	2007-07-26 15:24:28 UTC (rev 2398)
@@ -0,0 +1,17 @@
+---
+ sound/soc/s3c24xx/Kconfig |    2 	1 +	1 -	0 !
+ 1 file changed, 1 insertion(+), 1 deletion(-)
+
+Index: linux-2.6/sound/soc/s3c24xx/Kconfig
+===================================================================
+--- linux-2.6.orig/sound/soc/s3c24xx/Kconfig	2007-07-23 10:15:13.000000000 +0200
++++ linux-2.6/sound/soc/s3c24xx/Kconfig	2007-07-23 10:18:07.000000000 +0200
+@@ -18,7 +18,7 @@ config SND_S3C2443_SOC_AC97
+ 	
+ config SND_S3C24XX_SOC_NEO1973_WM8753
+ 	tristate "SoC I2S Audio support for NEO1973 - WM8753"
+-	depends on SND_S3C24XX_SOC && MACH_GTA01
++	depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
+ 	select SND_S3C24XX_SOC_I2S
+ 	select SND_SOC_WM8753
+ 	help

Deleted: branches/src/target/kernel/2.6.22.x/patches/asoc.patch
===================================================================
--- branches/src/target/kernel/2.6.22.x/patches/asoc.patch	2007-07-26 14:40:11 UTC (rev 2397)
+++ branches/src/target/kernel/2.6.22.x/patches/asoc.patch	2007-07-26 15:24:28 UTC (rev 2398)
@@ -1,24271 +0,0 @@
-Index: linux-2.6.21-moko/sound/soc/codecs/ak4535.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/ak4535.c
-@@ -0,0 +1,697 @@
-+/*
-+ * ak4535.c  --  AK4535 ALSA Soc Audio driver
-+ *
-+ * Copyright 2005 Openedhand Ltd.
-+ *
-+ * Author: Richard Purdie <richard at openedhand.com>
-+ *
-+ * Based on wm8753.c by Liam Girdwood
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "ak4535.h"
-+
-+#define AUDIO_NAME "ak4535"
-+#define AK4535_VERSION "0.3"
-+
-+struct snd_soc_codec_device soc_codec_dev_ak4535;
-+
-+/* codec private data */
-+struct ak4535_priv {
-+	unsigned int sysclk;
-+};
-+
-+/*
-+ * ak4535 register cache
-+ */
-+static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
-+    0x0000, 0x0080, 0x0000, 0x0003,
-+    0x0002, 0x0000, 0x0011, 0x0001,
-+    0x0000, 0x0040, 0x0036, 0x0010,
-+    0x0000, 0x0000, 0x0057, 0x0000,
-+};
-+
-+/*
-+ * read ak4535 register cache
-+ */
-+static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= AK4535_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write ak4535 register cache
-+ */
-+static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= AK4535_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the AK4535 register space
-+ */
-+static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D8 AK4535 register offset
-+	 *   D7...D0 register data
-+	 */
-+	data[0] = reg & 0xff;
-+	data[1] = value & 0xff;
-+
-+	ak4535_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
-+static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
-+static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
-+static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
-+static const char *ak4535_mic_select[] = {"Internal", "External"};
-+
-+static const struct soc_enum ak4535_enum[] = {
-+	SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
-+	SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
-+	SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
-+	SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
-+	SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
-+};
-+
-+static const struct snd_kcontrol_new ak4535_snd_controls[] = {
-+	SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
-+	SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
-+	SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
-+	SOC_ENUM("Headphone Output", ak4535_enum[2]),
-+	SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
-+	SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
-+	SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
-+	SOC_ENUM("Mic Select", ak4535_enum[4]),
-+	SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
-+	SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
-+	SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
-+	SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
-+	SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
-+	SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
-+	SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
-+	SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
-+	SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
-+	SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
-+	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
-+};
-+
-+/* add non dapm controls */
-+static int ak4535_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+			snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Mono 1 Mixer */
-+static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
-+	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
-+	SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
-+};
-+
-+/* Stereo Mixer */
-+static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
-+	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
-+	SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
-+	SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
-+};
-+
-+/* Input Mixer */
-+static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
-+	SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
-+	SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
-+};
-+
-+/* Input mux */
-+static const struct snd_kcontrol_new ak4535_input_mux_control =
-+	SOC_DAPM_ENUM("Input Select", ak4535_enum[0]);
-+
-+/* HP L switch */
-+static const struct snd_kcontrol_new ak4535_hpl_control =
-+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
-+
-+/* HP R switch */
-+static const struct snd_kcontrol_new ak4535_hpr_control =
-+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
-+
-+/* Speaker switch */
-+static const struct snd_kcontrol_new ak4535_spk_control =
-+	SOC_DAPM_SINGLE("Switch", AK4535_MODE2, 0, 0, 0);
-+
-+/* mono 2 switch */
-+static const struct snd_kcontrol_new ak4535_mono2_control =
-+	SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
-+
-+/* Line out switch */
-+static const struct snd_kcontrol_new ak4535_line_control =
-+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
-+
-+/* ak4535 dapm widgets */
-+static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
-+	SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
-+		&ak4535_stereo_mixer_controls[0],
-+		ARRAY_SIZE(ak4535_stereo_mixer_controls)),
-+	SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
-+		&ak4535_mono1_mixer_controls[0],
-+		ARRAY_SIZE(ak4535_mono1_mixer_controls)),
-+	SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
-+		&ak4535_input_mixer_controls[0],
-+		ARRAY_SIZE(ak4535_mono1_mixer_controls)),
-+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
-+		&ak4535_input_mux_control),
-+	SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
-+	SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
-+		&ak4535_mono2_control),
-+	SND_SOC_DAPM_SWITCH("Speaker Enable", SND_SOC_NOPM, 0, 0,
-+		&ak4535_spk_control),
-+	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
-+		&ak4535_line_control),
-+	SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
-+		&ak4535_hpl_control),
-+	SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
-+		&ak4535_hpr_control),
-+	SND_SOC_DAPM_OUTPUT("LOUT"),
-+	SND_SOC_DAPM_OUTPUT("HPL"),
-+	SND_SOC_DAPM_OUTPUT("ROUT"),
-+	SND_SOC_DAPM_OUTPUT("HPR"),
-+	SND_SOC_DAPM_OUTPUT("SPP"),
-+	SND_SOC_DAPM_OUTPUT("SPN"),
-+	SND_SOC_DAPM_OUTPUT("MOUT1"),
-+	SND_SOC_DAPM_OUTPUT("MOUT2"),
-+	SND_SOC_DAPM_OUTPUT("MICOUT"),
-+	SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 1),
-+	SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
-+
-+	SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
-+	SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
-+	SND_SOC_DAPM_INPUT("MICIN"),
-+	SND_SOC_DAPM_INPUT("MICEXT"),
-+	SND_SOC_DAPM_INPUT("AUX"),
-+	SND_SOC_DAPM_INPUT("MIN"),
-+	SND_SOC_DAPM_INPUT("AIN"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/*stereo mixer */
-+	{"Stereo Mixer", "Playback Switch", "DAC"},
-+	{"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
-+	{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
-+
-+	/* mono1 mixer */
-+	{"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
-+	{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
-+
-+	/* mono2 mixer */
-+	{"Mono2 Mixer", "Mono Playback Switch", "Stereo Mixer"},
-+
-+	/* Mic */
-+	{"AIN", NULL, "Mic"},
-+	{"Input Mux", "Internal", "Mic Int Bias"},
-+	{"Input Mux", "External", "Mic Ext Bias"},
-+	{"Mic Int Bias", NULL, "MICIN"},
-+	{"Mic Ext Bias", NULL, "MICEXT"},
-+	{"MICOUT", NULL, "Input Mux"},
-+
-+	/* line out */
-+	{"LOUT", "Switch", "Line"},
-+	{"ROUT", "Switch", "Line Out Enable"},
-+	{"Line Out Enable", NULL, "Line Out"},
-+	{"Line Out", NULL, "Stereo Mixer"},
-+
-+	/* mono1 out */
-+	{"MOUT1", NULL, "Mono Out"},
-+	{"Mono Out", NULL, "Mono Mixer"},
-+
-+	/* left HP */
-+	{"HPL", "Switch", "Left HP Enable"},
-+	{"Left HP Enable", NULL, "HP L Amp"},
-+	{"HP L Amp", NULL, "Stereo Mixer"},
-+
-+	/* right HP */
-+	{"HPR", "Switch", "Right HP Enable"},
-+	{"Right HP Enable", NULL, "HP R Amp"},
-+	{"HP R Amp", NULL, "Stereo Mixer"},
-+
-+	/* speaker */
-+	{"SPP", "Switch", "Speaker Enable"},
-+	{"SPN", "Switch", "Speaker Enable"},
-+	{"Speaker Enable", NULL, "Spk Amp"},
-+	{"Spk Amp", NULL, "MIN"},
-+
-+	/* mono 2 */
-+	{"MOUT2", "Switch", "Mono 2 Enable"},
-+	{"Mono 2 Enable", NULL, "Stereo Mixer"},
-+
-+	/* Aux In */
-+	{"Aux In", NULL, "AUX"},
-+
-+	/* ADC */
-+	{"ADC", NULL, "Input Mixer"},
-+	{"Input Mixer", "Mic Capture Switch", "Mic"},
-+	{"Input Mixer", "Aux Capture Switch", "Aux In"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int ak4535_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
-+	int clk_id, unsigned int freq, int dir)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	struct ak4535_priv *ak4535 = codec->private_data;
-+
-+	ak4535->sysclk = freq;
-+	return 0;
-+}
-+
-+static int ak4535_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct ak4535_priv *ak4535 = codec->private_data;
-+	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
-+	int rate = params_rate(params), fs = 256;
-+
-+	if (rate)
-+		fs = ak4535->sysclk / rate;
-+
-+	/* set fs */
-+	switch (fs) {
-+	case 1024:
-+		mode2 |= (0x2 << 5);
-+		break;
-+	case 512:
-+		mode2 |= (0x1 << 5);
-+		break;
-+	case 256:
-+		break;
-+	}
-+
-+	/* set rate */
-+	ak4535_write(codec, AK4535_MODE2, mode2);
-+	return 0;
-+}
-+
-+static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u8 mode1 = 0;
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		mode1 = 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		mode1 = 0x0001;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* use 32 fs for BCLK to save power */
-+	mode1 |= 0x4;
-+
-+	ak4535_write(codec, AK4535_MODE1, mode1);
-+	return 0;
-+}
-+
-+static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
-+	if (mute)
-+		ak4535_write(codec, AK4535_DAC, mute_reg);
-+	else
-+		ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
-+	return 0;
-+}
-+
-+static int ak4535_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+	/* vref/mid, clk and osc on, dac unmute, active */
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, dac mute, inactive */
-+		ak4535_write(codec, AK4535_PM1, 0x80);
-+		ak4535_write(codec, AK4535_PM2, 0x0);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, inactive */
-+		ak4535_write(codec, AK4535_PM1, 0x0);
-+		ak4535_write(codec, AK4535_PM2, 0x80);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000)
-+
-+struct snd_soc_codec_dai ak4535_dai = {
-+	.name = "AK4535",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = AK4535_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = AK4535_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.hw_params = ak4535_hw_params,
-+	},
-+	.dai_ops = {
-+		.set_fmt = ak4535_set_dai_fmt,
-+		.digital_mute = ak4535_mute,
-+		.set_sysclk = ak4535_set_dai_sysclk,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(ak4535_dai);
-+
-+static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int ak4535_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(ak4535_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	ak4535_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the AK4535 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int ak4535_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "AK4535";
-+	codec->owner = THIS_MODULE;
-+	codec->read = ak4535_read_reg_cache;
-+	codec->write = ak4535_write;
-+	codec->dapm_event = ak4535_dapm_event;
-+	codec->dai = &ak4535_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(ak4535_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, ak4535_reg,
-+		sizeof(u16) * ARRAY_SIZE(ak4535_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ak4535_reg);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "ak4535: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	ak4535_add_controls(codec);
-+	ak4535_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+		printk(KERN_ERR "ak4535: failed to register card\n");
-+		goto card_err;
-+	}
-+
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+
-+	return ret;
-+}
-+
-+static struct snd_soc_device *ak4535_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+#define I2C_DRIVERID_AK4535 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver ak4535_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = ak4535_socdev;
-+	struct ak4535_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = ak4535_init(socdev);
-+	if (ret < 0) {
-+		printk(KERN_ERR "failed to initialise AK4535\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int ak4535_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int ak4535_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, ak4535_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver ak4535_i2c_driver = {
-+	.driver = {
-+		.name = "AK4535 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_AK4535,
-+	.attach_adapter = ak4535_i2c_attach,
-+	.detach_client =  ak4535_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "AK4535",
-+	.driver = &ak4535_i2c_driver,
-+};
-+#endif
-+
-+static int ak4535_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct ak4535_setup_data *setup;
-+	struct snd_soc_codec* codec;
-+	struct ak4535_priv *ak4535;
-+	int ret = 0;
-+
-+	printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL);
-+	if (ak4535 == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+
-+	codec->private_data = ak4535;
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	ak4535_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&ak4535_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int ak4535_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec* codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&ak4535_i2c_driver);
-+#endif
-+	kfree(codec->private_data);
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_ak4535 = {
-+	.probe = 	ak4535_probe,
-+	.remove = 	ak4535_remove,
-+	.suspend = 	ak4535_suspend,
-+	.resume =	ak4535_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
-+
-+MODULE_DESCRIPTION("Soc AK4535 driver");
-+MODULE_AUTHOR("Richard Purdie");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/ak4535.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/ak4535.h
-@@ -0,0 +1,46 @@
-+/*
-+ * ak4535.h  --  AK4535 Soc Audio driver
-+ *
-+ * Copyright 2005 Openedhand Ltd.
-+ *
-+ * Author: Richard Purdie <richard at openedhand.com>
-+ *
-+ * Based on wm8753.h
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _AK4535_H
-+#define _AK4535_H
-+
-+/* AK4535 register space */
-+
-+#define AK4535_PM1		0x0
-+#define AK4535_PM2		0x1
-+#define AK4535_SIG1		0x2
-+#define AK4535_SIG2		0x3
-+#define AK4535_MODE1	0x4
-+#define AK4535_MODE2	0x5
-+#define AK4535_DAC		0x6
-+#define AK4535_MIC		0x7
-+#define AK4535_TIMER	0x8
-+#define AK4535_ALC1		0x9
-+#define AK4535_ALC2		0xa
-+#define AK4535_PGA		0xb
-+#define AK4535_LATT		0xc
-+#define AK4535_RATT		0xd
-+#define AK4535_VOL		0xe
-+#define AK4535_STATUS	0xf
-+
-+#define AK4535_CACHEREGNUM 	0x10
-+
-+struct ak4535_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai ak4535_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_ak4535;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/uda1380.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/uda1380.c
-@@ -0,0 +1,745 @@
-+/*
-+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ *
-+ * Modified by Richard Purdie <richard at openedhand.com> to fit into SoC
-+ * codec model.
-+ *
-+ * Copyright (c) 2005 Giorgio Padrin <giorgio at mandarinlogiq.org>
-+ * Copyright 2005 Openedhand Ltd.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/types.h>
-+#include <linux/string.h>
-+#include <linux/slab.h>
-+#include <linux/errno.h>
-+#include <linux/ioctl.h>
-+#include <linux/delay.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/control.h>
-+#include <sound/initval.h>
-+#include <sound/info.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include "uda1380.h"
-+
-+#define UDA1380_VERSION "0.5"
-+#define AUDIO_NAME "uda1380"
-+/*
-+ * Debug
-+ */
-+
-+#define UDA1380_DEBUG 0
-+
-+#ifdef UDA1380_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+
-+/*
-+ * uda1380 register cache
-+ */
-+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
-+	0x0502, 0x0000, 0x0000, 0x3f3f,
-+	0x0202, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0xff00, 0x0000, 0x4800,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x8000, 0x0002, 0x0000,
-+};
-+
-+/*
-+ * read uda1380 register cache
-+ */
-+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == UDA1380_RESET)
-+		return 0;
-+	if (reg >= UDA1380_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write uda1380 register cache
-+ */
-+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= UDA1380_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the UDA1380 register space
-+ */
-+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[3];
-+
-+	/* data is
-+	 *   data[0] is register offset
-+	 *   data[1] is MS byte
-+	 *   data[2] is LS byte
-+	 */
-+	data[0] = reg;
-+	data[1] = (value & 0xff00) >> 8;
-+	data[2] = value & 0x00ff;
-+
-+	uda1380_write_reg_cache (codec, reg, value);
-+
-+	/* the interpolator & decimator regs must only be written when the
-+	 * codec DAI is active.
-+	 */
-+	if (!codec->active && (reg >= UDA1380_MVOL))
-+		return 0;
-+	dbg("uda hw write %x val %x\n", reg, value);
-+	if (codec->hw_write(codec->control_data, data, 3) == 3) {
-+		unsigned int val;
-+		i2c_master_send(codec->control_data, data, 1);
-+		i2c_master_recv(codec->control_data, data, 2);
-+		val = (data[0]<<8) | data[1];
-+		if (val != value) {
-+			dbg("READ BACK VAL %x\n", (data[0]<<8) | data[1]);
-+			return -EIO;
-+		}
-+		return 0;
-+	} else
-+		return -EIO;
-+}
-+
-+#define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
-+
-+/* declarations of ALSA reg_elem_REAL controls */
-+static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz",
-+				      "96kHz"};
-+static const char *uda1380_input_sel[] = {"Line", "Mic"};
-+static const char *uda1380_output_sel[] = {"Direct", "Mixer"};
-+static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2",
-+					 "Maximum"};
-+
-+static const struct soc_enum uda1380_enum[] = {
-+	SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp),
-+	SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel),
-+	SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode),
-+	SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel), /* R02_EN_AVC */
-+};
-+
-+static const struct snd_kcontrol_new uda1380_snd_controls[] = {
-+	SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 255, 1),
-+	SOC_DOUBLE("Mixer Volume", UDA1380_MIXVOL, 0, 8, 255, 1),
-+	SOC_ENUM("Sound Processing Filter Mode", uda1380_enum[2]),
-+	SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0),
-+	SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0),
-+	SOC_ENUM("Playback De-emphasis", uda1380_enum[0]),
-+	SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0),
-+	SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0),
-+	SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0),
-+	SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 1),
-+	SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0),
-+	SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0),
-+	SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1),
-+	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
-+	SOC_SINGLE("Silence", UDA1380_MIXER, 7, 1, 0),
-+	SOC_SINGLE("Silence Detection", UDA1380_MIXER, 6, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int uda1380_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+			snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Input mux */
-+static const struct snd_kcontrol_new uda1380_input_mux_control =
-+	SOC_DAPM_ENUM("Input Select", uda1380_enum[1]);
-+
-+/* Output mux */
-+static const struct snd_kcontrol_new uda1380_output_mux_control =
-+	SOC_DAPM_ENUM("Output Select", uda1380_enum[3]);
-+
-+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
-+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
-+		&uda1380_input_mux_control),
-+	SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
-+		&uda1380_output_mux_control),
-+	SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
-+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
-+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
-+	SND_SOC_DAPM_INPUT("VINM"),
-+	SND_SOC_DAPM_INPUT("VINL"),
-+	SND_SOC_DAPM_INPUT("VINR"),
-+	SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
-+	SND_SOC_DAPM_OUTPUT("VOUTLHP"),
-+	SND_SOC_DAPM_OUTPUT("VOUTRHP"),
-+	SND_SOC_DAPM_OUTPUT("VOUTL"),
-+	SND_SOC_DAPM_OUTPUT("VOUTR"),
-+	SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
-+	SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
-+};
-+
-+static const char *audio_map[][3] = {
-+
-+	/* output mux */
-+	{"HeadPhone Driver", NULL, "Output Mux"},
-+	{"VOUTR", NULL, "Output Mux"},
-+	{"VOUTL", NULL, "Output Mux"},
-+
-+	{"Analog Mixer", NULL, "VINR"},
-+	{"Analog Mixer", NULL, "VINL"},
-+	{"Analog Mixer", NULL, "DAC"},
-+
-+	{"Output Mux", "Direct", "DAC"},
-+	{"Output Mux", "Mixer", "Analog Mixer"},
-+
-+	/* headphone driver */
-+	{"VOUTLHP", NULL, "HeadPhone Driver"},
-+	{"VOUTRHP", NULL, "HeadPhone Driver"},
-+
-+	/* input mux */
-+	{"Left ADC", NULL, "Input Mux"},
-+	{"Input Mux", "Mic", "Mic LNA"},
-+	{"Input Mux", "Line", "Left PGA"},
-+
-+	/* right input */
-+	{"Right ADC", NULL, "Right PGA"},
-+
-+	/* inputs */
-+	{"Mic LNA", NULL, "VINM"},
-+	{"Left PGA", NULL, "VINL"},
-+	{"Right PGA", NULL, "VINR"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int uda1380_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++)
-+		snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
-+
-+	/* set up audio path interconnects */
-+	for (i = 0; audio_map[i][0] != NULL; i++)
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	int iface;
-+	/* set up DAI based upon fmt */
-+
-+	iface = uda1380_read_reg_cache (codec, UDA1380_IFACE);
-+	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
-+
-+	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
-+		break;
-+	case SND_SOC_DAIFMT_LSB:
-+		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
-+		break;
-+	case SND_SOC_DAIFMT_MSB:
-+		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
-+	}
-+
-+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
-+		iface |= R01_SIM;
-+
-+	uda1380_write(codec, UDA1380_IFACE, iface);
-+
-+	return 0;
-+}
-+
-+/*
-+ * Flush reg cache
-+ * We can only write the interpolator and decimator registers
-+ * when the DAI is being clocked by the CPU DAI. It's up to the
-+ * machine and cpu DAI driver to do this before we are called.
-+ */
-+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, reg_start, reg_end, clk;
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		reg_start = UDA1380_MVOL;
-+		reg_end = UDA1380_MIXER;
-+	} else {
-+		reg_start = UDA1380_DEC;
-+		reg_end = UDA1380_AGC;
-+	}
-+
-+	/* FIXME disable DAC_CLK */
-+	clk = uda1380_read_reg_cache (codec, 00);
-+	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
-+
-+	for (reg = reg_start; reg <= reg_end; reg++ ) {
-+		dbg("reg %x val %x\n",reg, uda1380_read_reg_cache (codec, reg));
-+		uda1380_write(codec, reg, uda1380_read_reg_cache (codec, reg));
-+	}
-+
-+	/* FIXME enable DAC_CLK */
-+	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
-+
-+	return 0;
-+}
-+
-+static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-+
-+	/* set WSPLL power and divider if running from this clock */
-+	if (clk & R00_DAC_CLK) {
-+		int rate = params_rate(params);
-+		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
-+		clk &= ~0x3; /* clear SEL_LOOP_DIV */
-+		switch (rate) {
-+		case 6250 ... 12500:
-+			clk |= 0x0;
-+			break;
-+		case 12501 ... 25000:
-+			clk |= 0x1;
-+			break;
-+		case 25001 ... 50000:
-+			clk |= 0x2;
-+			break;
-+		case 50001 ... 100000:
-+			clk |= 0x3;
-+			break;
-+		}
-+		uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
-+	}
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+		clk |= R00_EN_DAC | R00_EN_INT;
-+	else
-+		clk |= R00_EN_ADC | R00_EN_DEC;
-+
-+	uda1380_write(codec, UDA1380_CLK, clk);
-+	return 0;
-+}
-+
-+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-+
-+	/* shut down WSPLL power if running from this clock */
-+	if (clk & R00_DAC_CLK) {
-+		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
-+		uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
-+	}
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+		clk &= ~(R00_EN_DAC | R00_EN_INT);
-+	else
-+		clk &= ~(R00_EN_ADC | R00_EN_DEC);
-+
-+	uda1380_write(codec, UDA1380_CLK, clk);
-+}
-+
-+static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff;
-+
-+	/* FIXME: mute(codec,0) is called when the magician clock is already
-+	 * set to WSPLL, but for some unknown reason writing to interpolator
-+	 * registers works only when clocked by SYSCLK */
-+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-+	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
-+	if (mute)
-+		uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000);
-+	else
-+		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
-+	uda1380_write(codec, UDA1380_CLK, clk);
-+	return 0;
-+}
-+
-+static int uda1380_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		/* enable internal bias */
-+		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except internal bias */
-+		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, inactive */
-+		uda1380_write(codec, UDA1380_PM, 0x0);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		       SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-+		       SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-+
-+struct snd_soc_codec_dai uda1380_dai[] = {
-+{
-+	.name = "UDA1380",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = UDA1380_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = UDA1380_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.hw_params = uda1380_pcm_hw_params,
-+		.shutdown = uda1380_pcm_shutdown,
-+		.prepare = uda1380_pcm_prepare,
-+	},
-+	.dai_ops = {
-+		.digital_mute = uda1380_mute,
-+		.set_fmt = uda1380_set_dai_fmt,
-+	},
-+},
-+{/* playback only  - dual interface */
-+	.name = "UDA1380",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = UDA1380_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-+	},
-+	.ops = {
-+		.hw_params = uda1380_pcm_hw_params,
-+		.shutdown = uda1380_pcm_shutdown,
-+		.prepare = uda1380_pcm_prepare,
-+	},
-+	.dai_ops = {
-+		.digital_mute = uda1380_mute,
-+		.set_fmt = uda1380_set_dai_fmt,
-+	},
-+},
-+{ /* capture only - dual interface*/
-+	.name = "UDA1380",
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = UDA1380_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-+	},
-+	.ops = {
-+		.hw_params = uda1380_pcm_hw_params,
-+		.shutdown = uda1380_pcm_shutdown,
-+		.prepare = uda1380_pcm_prepare,
-+	},
-+	.dai_ops = {
-+		.set_fmt = uda1380_set_dai_fmt,
-+	},
-+},
-+};
-+EXPORT_SYMBOL_GPL(uda1380_dai);
-+
-+static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int uda1380_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	uda1380_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the UDA1380 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "UDA1380";
-+	codec->owner = THIS_MODULE;
-+	codec->read = uda1380_read_reg_cache;
-+	codec->write = uda1380_write;
-+	codec->dapm_event = uda1380_dapm_event;
-+	codec->dai = uda1380_dai;
-+	codec->num_dai = ARRAY_SIZE(uda1380_dai);
-+	codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
-+	codec->reg_cache =
-+		kcalloc(ARRAY_SIZE(uda1380_reg), sizeof(u16), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, uda1380_reg,
-+	       sizeof(u16) * ARRAY_SIZE(uda1380_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(uda1380_reg);
-+	uda1380_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "uda1380: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	/* set clock input */
-+	switch (dac_clk) {
-+	case UDA1380_DAC_CLK_SYSCLK:
-+		uda1380_write(codec, UDA1380_CLK, 0);
-+		break;
-+	case UDA1380_DAC_CLK_WSPLL:
-+		uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
-+		break;
-+	}
-+
-+	/* uda1380 init */
-+	uda1380_add_controls(codec);
-+	uda1380_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+		printk(KERN_ERR "uda1380: failed to register card\n");
-+		goto card_err;
-+	}
-+
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *uda1380_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+#define I2C_DRIVERID_UDA1380 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver uda1380_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = uda1380_socdev;
-+	struct uda1380_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = uda1380_init(socdev, setup->dac_clk);
-+	if (ret < 0) {
-+		printk(KERN_ERR "failed to initialise UDA1380\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int uda1380_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int uda1380_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, uda1380_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver uda1380_i2c_driver = {
-+	.driver = {
-+		.name =  "UDA1380 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_UDA1380,
-+	.attach_adapter = uda1380_i2c_attach,
-+	.detach_client =  uda1380_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "UDA1380",
-+	.driver = &uda1380_i2c_driver,
-+};
-+#endif
-+
-+static int uda1380_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct uda1380_setup_data *setup;
-+	struct snd_soc_codec* codec;
-+	int ret = 0;
-+
-+	printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	uda1380_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&uda1380_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int uda1380_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec* codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&uda1380_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
-+	.probe = 	uda1380_probe,
-+	.remove = 	uda1380_remove,
-+	.suspend = 	uda1380_suspend,
-+	.resume =	uda1380_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
-+
-+MODULE_AUTHOR("Giorgio Padrin");
-+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/uda1380.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/uda1380.h
-@@ -0,0 +1,89 @@
-+/*
-+ * Audio support for Philips UDA1380
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ *
-+ * Copyright (c) 2005 Giorgio Padrin <giorgio at mandarinlogiq.org>
-+ */
-+
-+#ifndef _UDA1380_H
-+#define _UDA1380_H
-+
-+#define UDA1380_CLK	0x00
-+#define UDA1380_IFACE	0x01
-+#define UDA1380_PM	0x02
-+#define UDA1380_AMIX	0x03
-+#define UDA1380_HP	0x04
-+#define UDA1380_MVOL	0x10
-+#define UDA1380_MIXVOL	0x11
-+#define UDA1380_MODE	0x12
-+#define UDA1380_DEEMP	0x13
-+#define UDA1380_MIXER	0x14
-+#define UDA1380_INTSTAT	0x18
-+#define UDA1380_DEC	0x20
-+#define UDA1380_PGA	0x21
-+#define UDA1380_ADC	0x22
-+#define UDA1380_AGC	0x23
-+#define UDA1380_DECSTAT	0x28
-+#define UDA1380_RESET	0x7f
-+
-+#define UDA1380_CACHEREGNUM 0x24
-+
-+/* Register flags */
-+#define R00_EN_ADC	0x0800
-+#define R00_EN_DEC	0x0400
-+#define R00_EN_DAC	0x0200
-+#define R00_EN_INT	0x0100
-+#define R00_DAC_CLK	0x0010
-+#define R01_SFORI_I2S   0x0000
-+#define R01_SFORI_LSB16 0x0100
-+#define R01_SFORI_LSB18 0x0200
-+#define R01_SFORI_LSB20 0x0300
-+#define R01_SFORI_MSB   0x0500
-+#define R01_SFORI_MASK  0x0700
-+#define R01_SFORO_I2S   0x0000
-+#define R01_SFORO_LSB16 0x0001
-+#define R01_SFORO_LSB18 0x0002
-+#define R01_SFORO_LSB20 0x0003
-+#define R01_SFORO_LSB24 0x0004
-+#define R01_SFORO_MSB   0x0005
-+#define R01_SFORO_MASK  0x0007
-+#define R01_SEL_SOURCE  0x0040
-+#define R01_SIM		0x0010
-+#define R02_PON_PLL	0x8000
-+#define R02_PON_HP	0x2000
-+#define R02_PON_DAC	0x0400
-+#define R02_PON_BIAS	0x0100
-+#define R02_EN_AVC	0x0080
-+#define R02_PON_AVC	0x0040
-+#define R02_PON_LNA	0x0010
-+#define R02_PON_PGAL	0x0008
-+#define R02_PON_ADCL	0x0004
-+#define R02_PON_PGAR	0x0002
-+#define R02_PON_ADCR	0x0001
-+#define R13_MTM		0x4000
-+#define R14_SILENCE	0x0080
-+#define R14_SDET_ON	0x0040
-+#define R21_MT_ADC	0x8000
-+#define R22_SEL_LNA	0x0008
-+#define R22_SEL_MIC	0x0004
-+#define R22_SKIP_DCFIL	0x0002
-+#define R23_AGC_EN	0x0001
-+
-+struct uda1380_setup_data {
-+	unsigned short i2c_address;
-+	int            dac_clk;
-+#define UDA1380_DAC_CLK_SYSCLK 0
-+#define UDA1380_DAC_CLK_WSPLL  1
-+};
-+
-+#define UDA1380_DAI_DUPLEX	0 /* playback and capture on single DAI */
-+#define UDA1380_DAI_PLAYBACK	1 /* playback DAI */
-+#define UDA1380_DAI_CAPTURE	2 /* capture DAI */
-+
-+extern struct snd_soc_codec_dai uda1380_dai[3];
-+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
-+
-+#endif /* _UDA1380_H */
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8753.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8753.c
-@@ -0,0 +1,1782 @@
-+/*
-+ * wm8753.c  --  WM8753 ALSA Soc Audio driver
-+ *
-+ * Copyright 2003 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ * Notes:
-+ *  The WM8753 is a low power, high quality stereo codec with integrated PCM
-+ *  codec designed for portable digital telephony applications.
-+ *
-+ * Dual DAI:-
-+ *
-+ * This driver support 2 DAI PCM's. This makes the default PCM available for
-+ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for
-+ * voice.
-+ *
-+ * Please note that the voice PCM can be connected directly to a Bluetooth
-+ * codec or GSM modem and thus cannot be read or written to, although it is
-+ * available to be configured with snd_hw_params(), etc and kcontrols in the
-+ * normal alsa manner.
-+ *
-+ * Fast DAI switching:-
-+ *
-+ * The driver can now fast switch between the DAI configurations via a
-+ * an alsa kcontrol. This allows the PCM to remain open.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+#include <asm/div64.h>
-+
-+#include "wm8753.h"
-+
-+#define AUDIO_NAME "wm8753"
-+#define WM8753_VERSION "0.16"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8753_DEBUG 0
-+
-+#ifdef WM8753_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+static int caps_charge = 2000;
-+module_param(caps_charge, int, 0);
-+MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
-+
-+static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
-+	unsigned int mode);
-+
-+/* codec private data */
-+struct wm8753_priv {
-+	unsigned int sysclk;
-+	unsigned int pcmclk;
-+};
-+
-+/*
-+ * wm8753 register cache
-+ * We can't read the WM8753 register space when we
-+ * are using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8753_reg[] = {
-+	0x0008, 0x0000, 0x000a, 0x000a,
-+	0x0033, 0x0000, 0x0007, 0x00ff,
-+	0x00ff, 0x000f, 0x000f, 0x007b,
-+	0x0000, 0x0032, 0x0000, 0x00c3,
-+	0x00c3, 0x00c0, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0055,
-+	0x0005, 0x0050, 0x0055, 0x0050,
-+	0x0055, 0x0050, 0x0055, 0x0079,
-+	0x0079, 0x0079, 0x0079, 0x0079,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0097, 0x0097, 0x0000, 0x0004,
-+	0x0000, 0x0083, 0x0024, 0x01ba,
-+	0x0000, 0x0083, 0x0024, 0x01ba,
-+	0x0000, 0x0000
-+};
-+
-+/*
-+ * read wm8753 register cache
-+ */
-+static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
-+		return -1;
-+	return cache[reg - 1];
-+}
-+
-+/*
-+ * write wm8753 register cache
-+ */
-+static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg < 1 || reg > 0x3f)
-+		return;
-+	cache[reg - 1] = value;
-+}
-+
-+/*
-+ * write to the WM8753 register space
-+ */
-+static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8753 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8753_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0)
-+
-+/*
-+ * WM8753 Controls
-+ */
-+static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"};
-+static const char *wm8753_base_filter[] =
-+	{"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz",
-+	"100Hz @ 8kHz", "200Hz @ 8kHz"};
-+static const char *wm8753_treble[] = {"8kHz", "4kHz"};
-+static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"};
-+static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"};
-+static const char *wm8753_3d_func[] = {"Capture", "Playback"};
-+static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"};
-+static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"};
-+static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"};
-+static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"};
-+static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"};
-+static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2",
-+	"Line 1", "Line 2"};
-+static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"};
-+static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"};
-+static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"};
-+static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"};
-+static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2",
-+	"Right PGA"};
-+static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right",
-+	"Left + Right"};
-+static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"};
-+static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"};
-+static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"};
-+static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"};
-+static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left",
-+	"Analogue Mix Right", "Digital Mono Mix"};
-+static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k",
-+	"82Hz @ 8kHz", "170Hz @ 8kHz"};
-+static const char *wm8753_adc_filter[] = {"HiFi", "Voice"};
-+static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
-+static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
-+static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC",
-+	"Channel Swap"};
-+
-+static const struct soc_enum wm8753_enum[] = {
-+SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base),		// 0
-+SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter),	// 1
-+SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble), 	// 2
-+SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func),	// 3
-+SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type),	// 4
-+SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func),		// 5
-+SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc),			// 6
-+SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc),			// 7
-+SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp),		// 8
-+SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix),		// 9
-+SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase),	// 10
-+SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix),	// 11
-+SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux),	// 12
-+SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux),	// 13
-+SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux),	// 14
-+SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel),	// 15
-+SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),// 16
-+SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src),	// 17
-+SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3),		// 18
-+SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4),		// 19
-+SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel),	// 20
-+SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel),	// 21
-+SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc),	// 22
-+SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp),		// 23
-+SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter),	// 24
-+SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel),	// 25
-+SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), // 26
-+SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel),	// 27
-+};
-+
-+
-+static int wm8753_get_dai(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-+	int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
-+
-+	ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
-+	return 0;
-+}
-+
-+static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-+	int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
-+
-+	if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	mode &= 0xfff3;
-+	mode |= (ucontrol->value.integer.value[0] << 2);
-+
-+	wm8753_write(codec, WM8753_IOCTL, mode);
-+	wm8753_set_dai_mode(codec, ucontrol->value.integer.value[0]);
-+	return 1;
-+}
-+
-+static const struct snd_kcontrol_new wm8753_snd_controls[] = {
-+SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0),
-+
-+SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0),
-+
-+SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
-+SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
-+
-+SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
-+
-+SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
-+SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
-+SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
-+
-+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
-+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
-+
-+SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
-+SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-+SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-+SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
-+
-+SOC_ENUM("Bass Boost", wm8753_enum[0]),
-+SOC_ENUM("Bass Filter", wm8753_enum[1]),
-+SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
-+
-+SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
-+SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
-+
-+SOC_DOUBLE("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1),
-+SOC_SINGLE("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1),
-+
-+SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0),
-+SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
-+SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
-+
-+SOC_ENUM("Capture Filter Select", wm8753_enum[23]),
-+SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]),
-+SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1),
-+
-+SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0),
-+SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0),
-+SOC_ENUM("ALC Capture Function", wm8753_enum[3]),
-+SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0),
-+SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1),
-+SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1),
-+SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0),
-+SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0),
-+SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]),
-+SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0),
-+
-+SOC_ENUM("3D Function", wm8753_enum[5]),
-+SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]),
-+SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]),
-+SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0),
-+SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0),
-+
-+SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0),
-+SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0),
-+
-+SOC_ENUM("De-emphasis", wm8753_enum[8]),
-+SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
-+SOC_ENUM("Playback Phase", wm8753_enum[10]),
-+
-+SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
-+SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
-+
-+SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
-+
-+SOC_ENUM("ADC Data Select", wm8753_enum[27]),
-+};
-+
-+/* add non dapm controls */
-+static int wm8753_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+	return 0;
-+}
-+
-+/*
-+ * _DAPM_ Controls
-+ */
-+
-+/* Left Mixer */
-+static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0),
-+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0),
-+};
-+
-+/* Right mixer */
-+static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0),
-+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0),
-+};
-+
-+/* Mono mixer */
-+static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0),
-+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0),
-+};
-+
-+/* Mono 2 Mux */
-+static const struct snd_kcontrol_new wm8753_mono2_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[17]);
-+
-+/* Out 3 Mux */
-+static const struct snd_kcontrol_new wm8753_out3_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[18]);
-+
-+/* Out 4 Mux */
-+static const struct snd_kcontrol_new wm8753_out4_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[19]);
-+
-+/* ADC Mono Mix */
-+static const struct snd_kcontrol_new wm8753_adc_mono_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[22]);
-+
-+/* Record mixer */
-+static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0),
-+SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0),
-+SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0),
-+};
-+
-+/* Left ADC mux */
-+static const struct snd_kcontrol_new wm8753_adc_left_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[21]);
-+
-+/* Right ADC mux */
-+static const struct snd_kcontrol_new wm8753_adc_right_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[20]);
-+
-+/* MIC mux */
-+static const struct snd_kcontrol_new wm8753_mic_mux_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[16]);
-+
-+/* ALC mixer */
-+static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0),
-+SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0),
-+SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0),
-+SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0),
-+};
-+
-+/* Left Line mux */
-+static const struct snd_kcontrol_new wm8753_line_left_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[14]);
-+
-+/* Right Line mux */
-+static const struct snd_kcontrol_new wm8753_line_right_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[13]);
-+
-+/* Mono Line mux */
-+static const struct snd_kcontrol_new wm8753_line_mono_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[12]);
-+
-+/* Line mux and mixer */
-+static const struct snd_kcontrol_new wm8753_line_mux_mix_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[11]);
-+
-+/* Rx mux and mixer */
-+static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[15]);
-+
-+/* Mic Selector Mux */
-+static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls =
-+SOC_DAPM_ENUM("Route", wm8753_enum[25]);
-+
-+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
-+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0),
-+SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0,
-+	&wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)),
-+SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0),
-+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0),
-+SND_SOC_DAPM_OUTPUT("LOUT1"),
-+SND_SOC_DAPM_OUTPUT("LOUT2"),
-+SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0,
-+	&wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)),
-+SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0),
-+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0),
-+SND_SOC_DAPM_OUTPUT("ROUT1"),
-+SND_SOC_DAPM_OUTPUT("ROUT2"),
-+SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0,
-+	&wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)),
-+SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0),
-+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0),
-+SND_SOC_DAPM_OUTPUT("MONO1"),
-+SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls),
-+SND_SOC_DAPM_OUTPUT("MONO2"),
-+SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0),
-+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls),
-+SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0),
-+SND_SOC_DAPM_OUTPUT("OUT3"),
-+SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls),
-+SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0),
-+SND_SOC_DAPM_OUTPUT("OUT4"),
-+SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0,
-+	&wm8753_record_mixer_controls[0],
-+	ARRAY_SIZE(wm8753_record_mixer_controls)),
-+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8753_PWR2, 3, 0),
-+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8753_PWR2, 2, 0),
-+SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8753_adc_mono_controls),
-+SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8753_adc_mono_controls),
-+SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_adc_left_controls),
-+SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_adc_right_controls),
-+SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_mic_mux_controls),
-+SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0),
-+SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0,
-+	&wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)),
-+SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_line_left_controls),
-+SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_line_right_controls),
-+SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_line_mono_controls),
-+SND_SOC_DAPM_MUX("Line Mixer", WM8753_PWR2, 0, 0,
-+	&wm8753_line_mux_mix_controls),
-+SND_SOC_DAPM_MUX("Rx Mixer", WM8753_PWR2, 1, 0,
-+	&wm8753_rx_mux_mix_controls),
-+SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0),
-+SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0,
-+	&wm8753_mic_sel_mux_controls),
-+SND_SOC_DAPM_INPUT("LINE1"),
-+SND_SOC_DAPM_INPUT("LINE2"),
-+SND_SOC_DAPM_INPUT("RXP"),
-+SND_SOC_DAPM_INPUT("RXN"),
-+SND_SOC_DAPM_INPUT("ACIN"),
-+SND_SOC_DAPM_OUTPUT("ACOP"),
-+SND_SOC_DAPM_INPUT("MIC1N"),
-+SND_SOC_DAPM_INPUT("MIC1"),
-+SND_SOC_DAPM_INPUT("MIC2N"),
-+SND_SOC_DAPM_INPUT("MIC2"),
-+SND_SOC_DAPM_VMID("VREF"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* left mixer */
-+	{"Left Mixer", "Left Playback Switch", "Left DAC"},
-+	{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
-+	{"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
-+	{"Left Mixer", "Bypass Playback Switch", "Line Left Mux"},
-+
-+	/* right mixer */
-+	{"Right Mixer", "Right Playback Switch", "Right DAC"},
-+	{"Right Mixer", "Voice Playback Switch", "Voice DAC"},
-+	{"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
-+	{"Right Mixer", "Bypass Playback Switch", "Line Right Mux"},
-+
-+	/* mono mixer */
-+	{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
-+	{"Mono Mixer", "Left Playback Switch", "Left DAC"},
-+	{"Mono Mixer", "Right Playback Switch", "Right DAC"},
-+	{"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
-+	{"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"},
-+
-+	/* left out */
-+	{"Left Out 1", NULL, "Left Mixer"},
-+	{"Left Out 2", NULL, "Left Mixer"},
-+	{"LOUT1", NULL, "Left Out 1"},
-+	{"LOUT2", NULL, "Left Out 2"},
-+
-+	/* right out */
-+	{"Right Out 1", NULL, "Right Mixer"},
-+	{"Right Out 2", NULL, "Right Mixer"},
-+	{"ROUT1", NULL, "Right Out 1"},
-+	{"ROUT2", NULL, "Right Out 2"},
-+
-+	/* mono 1 out */
-+	{"Mono Out 1", NULL, "Mono Mixer"},
-+	{"MONO1", NULL, "Mono Out 1"},
-+
-+	/* mono 2 out */
-+	{"Mono 2 Mux", "Left + Right", "Out3 Left + Right"},
-+	{"Mono 2 Mux", "Inverted Mono 1", "MONO1"},
-+	{"Mono 2 Mux", "Left", "Left Mixer"},
-+	{"Mono 2 Mux", "Right", "Right Mixer"},
-+	{"Mono Out 2", NULL, "Mono 2 Mux"},
-+	{"MONO2", NULL, "Mono Out 2"},
-+
-+	/* out 3 */
-+	{"Out3 Left + Right", NULL, "Left Mixer"},
-+	{"Out3 Left + Right", NULL, "Right Mixer"},
-+	{"Out3 Mux", "VREF", "VREF"},
-+	{"Out3 Mux", "Left + Right", "Out3 Left + Right"},
-+	{"Out3 Mux", "ROUT2", "ROUT2"},
-+	{"Out 3", NULL, "Out3 Mux"},
-+	{"OUT3", NULL, "Out 3"},
-+
-+	/* out 4 */
-+	{"Out4 Mux", "VREF", "VREF"},
-+	{"Out4 Mux", "Capture ST", "Capture ST Mixer"},
-+	{"Out4 Mux", "LOUT2", "LOUT2"},
-+	{"Out 4", NULL, "Out4 Mux"},
-+	{"OUT4", NULL, "Out 4"},
-+
-+	/* record mixer  */
-+	{"Playback Mixer", "Left Capture Switch", "Left Mixer"},
-+	{"Playback Mixer", "Voice Capture Switch", "Mono Mixer"},
-+	{"Playback Mixer", "Right Capture Switch", "Right Mixer"},
-+
-+	/* Mic/SideTone Mux */
-+	{"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"},
-+	{"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"},
-+	{"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"},
-+	{"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"},
-+
-+	/* Capture Left Mux */
-+	{"Capture Left Mux", "PGA", "Left Capture Volume"},
-+	{"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"},
-+	{"Capture Left Mux", "Line", "LINE1"},
-+
-+	/* Capture Right Mux */
-+	{"Capture Right Mux", "PGA", "Right Capture Volume"},
-+	{"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
-+	{"Capture Right Mux", "Sidetone", "Capture ST Mixer"},
-+
-+	/* Mono Capture mixer-mux */
-+	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
-+	{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
-+	{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
-+	{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
-+	{"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"},
-+	{"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"},
-+	{"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"},
-+	{"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"},
-+	{"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"},
-+
-+	/* ADC */
-+	{"Left ADC", NULL, "Capture Left Mixer"},
-+	{"Right ADC", NULL, "Capture Right Mixer"},
-+
-+	/* Left Capture Volume */
-+	{"Left Capture Volume", NULL, "ACIN"},
-+
-+	/* Right Capture Volume */
-+	{"Right Capture Volume", NULL, "Mic 2 Volume"},
-+
-+	/* ALC Mixer */
-+	{"ALC Mixer", "Line Capture Switch", "Line Mixer"},
-+	{"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"},
-+	{"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"},
-+	{"ALC Mixer", "Rx Capture Switch", "Rx Mixer"},
-+
-+	/* Line Left Mux */
-+	{"Line Left Mux", "Line 1", "LINE1"},
-+	{"Line Left Mux", "Rx Mix", "Rx Mixer"},
-+
-+	/* Line Right Mux */
-+	{"Line Right Mux", "Line 2", "LINE2"},
-+	{"Line Right Mux", "Rx Mix", "Rx Mixer"},
-+
-+	/* Line Mono Mux */
-+	{"Line Mono Mux", "Line Mix", "Line Mixer"},
-+	{"Line Mono Mux", "Rx Mix", "Rx Mixer"},
-+
-+	/* Line Mixer/Mux */
-+	{"Line Mixer", "Line 1 + 2", "LINE1"},
-+	{"Line Mixer", "Line 1 - 2", "LINE1"},
-+	{"Line Mixer", "Line 1 + 2", "LINE2"},
-+	{"Line Mixer", "Line 1 - 2", "LINE2"},
-+	{"Line Mixer", "Line 1", "LINE1"},
-+	{"Line Mixer", "Line 2", "LINE2"},
-+
-+	/* Rx Mixer/Mux */
-+	{"Rx Mixer", "RXP - RXN", "RXP"},
-+	{"Rx Mixer", "RXP + RXN", "RXP"},
-+	{"Rx Mixer", "RXP - RXN", "RXN"},
-+	{"Rx Mixer", "RXP + RXN", "RXN"},
-+	{"Rx Mixer", "RXP", "RXP"},
-+	{"Rx Mixer", "RXN", "RXN"},
-+
-+	/* Mic 1 Volume */
-+	{"Mic 1 Volume", NULL, "MIC1N"},
-+	{"Mic 1 Volume", NULL, "Mic Selection Mux"},
-+
-+	/* Mic 2 Volume */
-+	{"Mic 2 Volume", NULL, "MIC2N"},
-+	{"Mic 2 Volume", NULL, "MIC2"},
-+
-+	/* Mic Selector Mux */
-+	{"Mic Selection Mux", "Mic 1", "MIC1"},
-+	{"Mic Selection Mux", "Mic 2", "MIC2N"},
-+	{"Mic Selection Mux", "Mic 3", "MIC2"},
-+
-+	/* ACOP */
-+	{"ACOP", NULL, "ALC Mixer"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8753_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-+		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-+
-+	/* set up the WM8753 audio map */
-+	for (i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+/* PLL divisors */
-+struct _pll_div {
-+	u32 div2:1;
-+	u32 n:4;
-+	u32 k:24;
-+};
-+
-+/* The size in bits of the pll divide multiplied by 10
-+ * to allow rounding later */
-+#define FIXED_PLL_SIZE ((1 << 22) * 10)
-+
-+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
-+	unsigned int source)
-+{
-+	unsigned long long Kpart;
-+	unsigned int K, Ndiv, Nmod;
-+
-+	Ndiv = target / source;
-+	if (Ndiv < 6) {
-+		source >>= 1;
-+		pll_div->div2 = 1;
-+		Ndiv = target / source;
-+	} else
-+		pll_div->div2 = 0;
-+
-+	if ((Ndiv < 6) || (Ndiv > 12))
-+		printk(KERN_WARNING
-+			"WM8753 N value outwith recommended range! N = %d\n",Ndiv);
-+
-+	pll_div->n = Ndiv;
-+	Nmod = target % source;
-+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
-+
-+	do_div(Kpart, source);
-+
-+	K = Kpart & 0xFFFFFFFF;
-+
-+	/* Check if we need to round */
-+	if ((K % 10) >= 5)
-+		K += 5;
-+
-+	/* Move down to proper range now rounding is done */
-+	K /= 10;
-+
-+	pll_div->k = K;
-+}
-+
-+static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	u16 reg, enable;
-+	int offset;
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+
-+	if (pll_id < WM8753_PLL1 || pll_id > WM8753_PLL2)
-+		return -ENODEV;
-+
-+	if (pll_id == WM8753_PLL1) {
-+		offset = 0;
-+		enable = 0x10;
-+		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef;
-+	} else {
-+		offset = 4;
-+		enable = 0x8;
-+		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7;
-+	}
-+
-+	if (!freq_in || !freq_out) {
-+		/* disable PLL  */
-+		wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0026);
-+		wm8753_write(codec, WM8753_CLOCK, reg);
-+		return 0;
-+	} else {
-+
-+        u16 value = 0;
-+        struct _pll_div pll_div;
-+
-+		pll_factors(&pll_div, freq_out * 8, freq_in);
-+
-+        /* set up N and K PLL divisor ratios */
-+        /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
-+        value = (pll_div.n << 5) + ((pll_div.k & 0x3c0000) >> 18);
-+        wm8753_write(codec, WM8753_PLL1CTL2 + offset, value);
-+
-+        /* bits 8:0 = PLL_K[17:9] */
-+        value = (pll_div.k & 0x03fe00) >> 9;
-+        wm8753_write(codec, WM8753_PLL1CTL3 + offset, value);
-+
-+        /* bits 8:0 = PLL_K[8:0] */
-+        value = pll_div.k & 0x0001ff;
-+        wm8753_write(codec, WM8753_PLL1CTL4 + offset, value);
-+
-+        /* set PLL as input and enable */
-+        wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0027 |
-+        	(pll_div.div2 << 3));
-+		wm8753_write(codec, WM8753_CLOCK, reg | enable);
-+	}
-+	return 0;
-+}
-+
-+struct _coeff_div {
-+	u32 mclk;
-+	u32 rate;
-+	u8 sr:5;
-+	u8 usb:1;
-+};
-+
-+/* codec hifi mclk (after PLL) clock divider coefficients */
-+static const struct _coeff_div coeff_div[] = {
-+	/* 8k */
-+	{12288000, 8000, 0x6, 0x0},
-+	{11289600, 8000, 0x16, 0x0},
-+	{18432000, 8000, 0x7, 0x0},
-+	{16934400, 8000, 0x17, 0x0},
-+	{12000000, 8000, 0x6, 0x1},
-+
-+	/* 11.025k */
-+	{11289600, 11025, 0x18, 0x0},
-+	{16934400, 11025, 0x19, 0x0},
-+	{12000000, 11025, 0x19, 0x1},
-+
-+	/* 16k */
-+	{12288000, 16000, 0xa, 0x0},
-+	{18432000, 16000, 0xb, 0x0},
-+	{12000000, 16000, 0xa, 0x1},
-+
-+	/* 22.05k */
-+	{11289600, 22050, 0x1a, 0x0},
-+	{16934400, 22050, 0x1b, 0x0},
-+	{12000000, 22050, 0x1b, 0x1},
-+
-+	/* 32k */
-+	{12288000, 32000, 0xc, 0x0},
-+	{18432000, 32000, 0xd, 0x0},
-+	{12000000, 32000, 0xa, 0x1},
-+
-+	/* 44.1k */
-+	{11289600, 44100, 0x10, 0x0},
-+	{16934400, 44100, 0x11, 0x0},
-+	{12000000, 44100, 0x11, 0x1},
-+
-+	/* 48k */
-+	{12288000, 48000, 0x0, 0x0},
-+	{18432000, 48000, 0x1, 0x0},
-+	{12000000, 48000, 0x0, 0x1},
-+
-+	/* 88.2k */
-+	{11289600, 88200, 0x1e, 0x0},
-+	{16934400, 88200, 0x1f, 0x0},
-+	{12000000, 88200, 0x1f, 0x1},
-+
-+	/* 96k */
-+	{12288000, 96000, 0xe, 0x0},
-+	{18432000, 96000, 0xf, 0x0},
-+	{12000000, 96000, 0xe, 0x1},
-+};
-+
-+static int get_coeff(int mclk, int rate)
-+{
-+	int i;
-+
-+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
-+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
-+			return i;
-+	}
-+	return -EINVAL;
-+}
-+
-+/*
-+ * Clock after PLL and dividers
-+ */
-+static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
-+		int clk_id, unsigned int freq, int dir)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	struct wm8753_priv *wm8753 = codec->private_data;
-+
-+	switch (freq) {
-+	case 11289600:
-+	case 12000000:
-+	case 12288000:
-+	case 16934400:
-+	case 18432000:
-+		if (clk_id == WM8753_MCLK) {
-+			wm8753->sysclk = freq;
-+			return 0;
-+		} else if (clk_id == WM8753_PCMCLK) {
-+			wm8753->pcmclk = freq;
-+			return 0;
-+		}
-+		break;
-+	}
-+	return -EINVAL;
-+}
-+
-+/*
-+ * Set's ADC and Voice DAC format.
-+ */
-+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01ec;
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		voice |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		voice |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		voice |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		voice |= 0x0013;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8753_write(codec, WM8753_PCM, voice);
-+	return 0;
-+}
-+
-+/*
-+ * Set PCM DAI bit size and sample rate.
-+ */
-+static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm8753_priv *wm8753 = codec->private_data;
-+	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
-+	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		voice |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		voice |= 0x0008;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		voice |= 0x000c;
-+		break;
-+	}
-+
-+	/* sample rate */
-+	if (params_rate(params) * 384 == wm8753->pcmclk)
-+		srate |= 0x80;
-+	wm8753_write(codec, WM8753_SRATE1, srate);
-+
-+	wm8753_write(codec, WM8753_PCM, voice);
-+	return 0;
-+}
-+
-+/*
-+ * Set's PCM dai fmt and BCLK.
-+ */
-+static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 voice, ioctl;
-+
-+	voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x011f;
-+	ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x015d;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		ioctl |= 0x2;
-+	case SND_SOC_DAIFMT_CBM_CFS:
-+		voice |= 0x0040;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_DSP_A:
-+	case SND_SOC_DAIFMT_DSP_B:
-+		/* frame inversion not valid for DSP modes */
-+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+		case SND_SOC_DAIFMT_NB_NF:
-+			break;
-+		case SND_SOC_DAIFMT_IB_NF:
-+			voice |= 0x0080;
-+			break;
-+		default:
-+			return -EINVAL;
-+		}
-+		break;
-+	case SND_SOC_DAIFMT_I2S:
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		voice &= ~0x0010;
-+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+		case SND_SOC_DAIFMT_NB_NF:
-+			break;
-+		case SND_SOC_DAIFMT_IB_IF:
-+			voice |= 0x0090;
-+			break;
-+		case SND_SOC_DAIFMT_IB_NF:
-+			voice |= 0x0080;
-+			break;
-+		case SND_SOC_DAIFMT_NB_IF:
-+			voice |= 0x0010;
-+			break;
-+		default:
-+			return -EINVAL;
-+		}
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8753_write(codec, WM8753_PCM, voice);
-+	wm8753_write(codec, WM8753_IOCTL, ioctl);
-+	return 0;
-+}
-+
-+static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8753_PCMDIV:
-+		reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f;
-+		wm8753_write(codec, WM8753_CLOCK, reg | div);
-+		break;
-+	case WM8753_BCLKDIV:
-+		reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7;
-+		wm8753_write(codec, WM8753_SRATE2, reg | div);
-+		break;
-+	case WM8753_VXCLKDIV:
-+		reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f;
-+		wm8753_write(codec, WM8753_SRATE2, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+	return 0;
-+}
-+
-+/*
-+ * Set's HiFi DAC format.
-+ */
-+static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0;
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		hifi |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		hifi |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		hifi |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		hifi |= 0x0013;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8753_write(codec, WM8753_HIFI, hifi);
-+	return 0;
-+}
-+
-+/*
-+ * Set's I2S DAI format.
-+ */
-+static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 ioctl, hifi;
-+
-+	hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x011f;
-+	ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00ae;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		ioctl |= 0x1;
-+	case SND_SOC_DAIFMT_CBM_CFS:
-+		hifi |= 0x0040;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_DSP_A:
-+	case SND_SOC_DAIFMT_DSP_B:
-+		/* frame inversion not valid for DSP modes */
-+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+		case SND_SOC_DAIFMT_NB_NF:
-+			break;
-+		case SND_SOC_DAIFMT_IB_NF:
-+			hifi |= 0x0080;
-+			break;
-+		default:
-+			return -EINVAL;
-+		}
-+		break;
-+	case SND_SOC_DAIFMT_I2S:
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		hifi &= ~0x0010;
-+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+		case SND_SOC_DAIFMT_NB_NF:
-+			break;
-+		case SND_SOC_DAIFMT_IB_IF:
-+			hifi |= 0x0090;
-+			break;
-+		case SND_SOC_DAIFMT_IB_NF:
-+			hifi |= 0x0080;
-+			break;
-+		case SND_SOC_DAIFMT_NB_IF:
-+			hifi |= 0x0010;
-+			break;
-+		default:
-+			return -EINVAL;
-+		}
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8753_write(codec, WM8753_HIFI, hifi);
-+	wm8753_write(codec, WM8753_IOCTL, ioctl);
-+	return 0;
-+}
-+
-+/*
-+ * Set PCM DAI bit size and sample rate.
-+ */
-+static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm8753_priv *wm8753 = codec->private_data;
-+	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
-+	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
-+	int coeff;
-+
-+	/* is digital filter coefficient valid ? */
-+	coeff = get_coeff(wm8753->sysclk, params_rate(params));
-+	if (coeff < 0) {
-+		printk(KERN_ERR "wm8753 invalid MCLK or rate\n");
-+		return coeff;
-+	}
-+	wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[coeff].sr << 1) |
-+		coeff_div[coeff].usb);
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		hifi |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		hifi |= 0x0008;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		hifi |= 0x000c;
-+		break;
-+	}
-+
-+	wm8753_write(codec, WM8753_HIFI, hifi);
-+	return 0;
-+}
-+
-+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 clock;
-+
-+	/* set clk source as pcmclk */
-+	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
-+	wm8753_write(codec, WM8753_CLOCK, clock);
-+
-+	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-+		return -EINVAL;
-+	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
-+}
-+
-+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
-+		return -EINVAL;
-+	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
-+}
-+
-+static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 clock;
-+
-+	/* set clk source as pcmclk */
-+	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
-+	wm8753_write(codec, WM8753_CLOCK, clock);
-+
-+	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-+		return -EINVAL;
-+	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
-+}
-+
-+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 clock;
-+
-+	/* set clk source as mclk */
-+	clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
-+	wm8753_write(codec, WM8753_CLOCK, clock | 0x4);
-+
-+	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
-+		return -EINVAL;
-+	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-+		return -EINVAL;
-+	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
-+}
-+
-+static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
-+
-+	/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
-+	 * make sure we check if they are not both active when we mute */
-+	if (mute && dai->id == 1) {
-+		if (!wm8753_dai[WM8753_DAI_VOICE].playback.active ||
-+			!wm8753_dai[WM8753_DAI_HIFI].playback.active)
-+			wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
-+	} else {
-+		if (mute)
-+			wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
-+		else
-+			wm8753_write(codec, WM8753_DAC, mute_reg);
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* set vmid to 50k and unmute dac */
-+		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		/* set vmid to 5k for quick power up */
-+		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* mute dac and set vmid to 500k, enable VREF */
-+		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		wm8753_write(codec, WM8753_PWR1, 0x0001);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+#define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+/*
-+ * The WM8753 supports upto 4 different and mutually exclusive DAI
-+ * configurations. This gives 2 PCM's available for use, hifi and voice.
-+ * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
-+ * is connected between the wm8753 and a BT codec or GSM modem.
-+ *
-+ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI
-+ * 2. Voice over HIFI DAI - HIFI disabled
-+ * 3. Voice disabled - HIFI over HIFI
-+ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
-+ */
-+static const struct snd_soc_codec_dai wm8753_all_dai[] = {
-+/* DAI HiFi mode 1 */
-+{	.name = "WM8753 HiFi",
-+	.id = 1,
-+	.playback = {
-+		.stream_name = "HiFi Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.capture = { /* dummy for fast DAI switching */
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8753_i2s_hw_params,},
-+	.dai_ops = {
-+		.digital_mute = wm8753_mute,
-+		.set_fmt = wm8753_mode1h_set_dai_fmt,
-+		.set_clkdiv = wm8753_set_dai_clkdiv,
-+		.set_pll = wm8753_set_dai_pll,
-+		.set_sysclk = wm8753_set_dai_sysclk,
-+	},
-+},
-+/* DAI Voice mode 1 */
-+{	.name = "WM8753 Voice",
-+	.id = 1,
-+	.playback = {
-+		.stream_name = "Voice Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8753_pcm_hw_params,},
-+	.dai_ops = {
-+		.digital_mute = wm8753_mute,
-+		.set_fmt = wm8753_mode1v_set_dai_fmt,
-+		.set_clkdiv = wm8753_set_dai_clkdiv,
-+		.set_pll = wm8753_set_dai_pll,
-+		.set_sysclk = wm8753_set_dai_sysclk,
-+	},
-+},
-+/* DAI HiFi mode 2 - dummy */
-+{	.name = "WM8753 HiFi",
-+	.id = 2,
-+},
-+/* DAI Voice mode 2 */
-+{	.name = "WM8753 Voice",
-+	.id = 2,
-+	.playback = {
-+		.stream_name = "Voice Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8753_pcm_hw_params,},
-+	.dai_ops = {
-+		.digital_mute = wm8753_mute,
-+		.set_fmt = wm8753_mode2_set_dai_fmt,
-+		.set_clkdiv = wm8753_set_dai_clkdiv,
-+		.set_pll = wm8753_set_dai_pll,
-+		.set_sysclk = wm8753_set_dai_sysclk,
-+	},
-+},
-+/* DAI HiFi mode 3 */
-+{	.name = "WM8753 HiFi",
-+	.id = 3,
-+	.playback = {
-+		.stream_name = "HiFi Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8753_i2s_hw_params,},
-+	.dai_ops = {
-+		.digital_mute = wm8753_mute,
-+		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-+		.set_clkdiv = wm8753_set_dai_clkdiv,
-+		.set_pll = wm8753_set_dai_pll,
-+		.set_sysclk = wm8753_set_dai_sysclk,
-+	},
-+},
-+/* DAI Voice mode 3 - dummy */
-+{	.name = "WM8753 Voice",
-+	.id = 3,
-+},
-+/* DAI HiFi mode 4 */
-+{	.name = "WM8753 HiFi",
-+	.id = 4,
-+	.playback = {
-+		.stream_name = "HiFi Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8753_RATES,
-+		.formats = WM8753_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8753_i2s_hw_params,},
-+	.dai_ops = {
-+		.digital_mute = wm8753_mute,
-+		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-+		.set_clkdiv = wm8753_set_dai_clkdiv,
-+		.set_pll = wm8753_set_dai_pll,
-+		.set_sysclk = wm8753_set_dai_sysclk,
-+	},
-+},
-+/* DAI Voice mode 4 - dummy */
-+{	.name = "WM8753 Voice",
-+	.id = 4,
-+},
-+};
-+
-+struct snd_soc_codec_dai wm8753_dai[2];
-+EXPORT_SYMBOL_GPL(wm8753_dai);
-+
-+static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
-+{
-+	if (mode < 4) {
-+		wm8753_dai[0] = wm8753_all_dai[mode << 1];
-+		wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
-+	}
-+	wm8753_dai[0].codec = codec;
-+	wm8753_dai[1].codec = codec;
-+}
-+
-+static void wm8753_work(struct work_struct *work)
-+{
-+	struct snd_soc_codec *codec =
-+		container_of(work, struct snd_soc_codec, delayed_work.work);
-+	wm8753_dapm_event(codec, codec->dapm_state);
-+}
-+
-+static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8753_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
-+		if (i + 1 == WM8753_RESET)
-+			continue;
-+		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+
-+	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* charge wm8753 caps */
-+	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-+		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-+		codec->dapm_state = SNDRV_CTL_POWER_D0;
-+		schedule_delayed_work(&codec->delayed_work,
-+			msecs_to_jiffies(caps_charge));
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8753 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8753_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "WM8753";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8753_read_reg_cache;
-+	codec->write = wm8753_write;
-+	codec->dapm_event = wm8753_dapm_event;
-+	codec->dai = wm8753_dai;
-+	codec->num_dai = 2;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
-+
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8753_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8753_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8753_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8753_reg);
-+	wm8753_set_dai_mode(codec, 0);
-+
-+	wm8753_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "wm8753: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* charge output caps */
-+	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-+	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
-+	schedule_delayed_work(&codec->delayed_work,
-+		msecs_to_jiffies(caps_charge));
-+
-+	/* set the update bits */
-+	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
-+	wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
-+	wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
-+	wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
-+	wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
-+	wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
-+	wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
-+	wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
-+	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
-+	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
-+
-+	wm8753_add_controls(codec);
-+	wm8753_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8753: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+static struct snd_soc_device *wm8753_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8753 2 wire address is determined by GPIO5
-+ * state during powerup.
-+ *    low  = 0x1a
-+ *    high = 0x1b
-+ */
-+#define I2C_DRIVERID_WM8753 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8753_i2c_driver;
-+static struct i2c_client client_template;
-+
-+static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8753_socdev;
-+	struct wm8753_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8753_init(socdev);
-+	if (ret < 0) {
-+		err("failed to initialise WM8753\n");
-+		goto err;
-+	}
-+
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int wm8753_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8753_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8753_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8753_i2c_driver = {
-+	.driver = {
-+		.name = "WM8753 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8753,
-+	.attach_adapter = wm8753_i2c_attach,
-+	.detach_client =  wm8753_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8753",
-+	.driver = &wm8753_i2c_driver,
-+};
-+#endif
-+
-+static int wm8753_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8753_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	struct wm8753_priv *wm8753;
-+	int ret = 0;
-+
-+	info("WM8753 Audio Codec %s", WM8753_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
-+	if (wm8753 == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+
-+	codec->private_data = wm8753;
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+	wm8753_socdev = socdev;
-+	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8753_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+		/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/*
-+ * This function forces any delayed work to be queued and run.
-+ */
-+static int run_delayed_work(struct delayed_work *dwork)
-+{
-+	int ret;
-+
-+	/* cancel any work waiting to be queued. */
-+	ret = cancel_delayed_work(dwork);
-+
-+	/* if there was any work waiting then we run it now and
-+	 * wait for it's completion */
-+	if (ret) {
-+		schedule_delayed_work(dwork, 0);
-+		flush_scheduled_work();
-+	}
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8753_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	run_delayed_work(&codec->delayed_work);
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8753_i2c_driver);
-+#endif
-+	kfree(codec->private_data);
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8753 = {
-+	.probe = 	wm8753_probe,
-+	.remove = 	wm8753_remove,
-+	.suspend = 	wm8753_suspend,
-+	.resume =	wm8753_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
-+
-+MODULE_DESCRIPTION("ASoC WM8753 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8753.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8753.h
-@@ -0,0 +1,126 @@
-+/*
-+ * wm8753.h  --  audio driver for WM8753
-+ *
-+ * Copyright 2003 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ */
-+
-+#ifndef _WM8753_H
-+#define _WM8753_H
-+
-+/* WM8753 register space */
-+
-+#define WM8753_DAC		0x01
-+#define WM8753_ADC		0x02
-+#define WM8753_PCM		0x03
-+#define WM8753_HIFI		0x04
-+#define WM8753_IOCTL		0x05
-+#define WM8753_SRATE1		0x06
-+#define WM8753_SRATE2		0x07
-+#define WM8753_LDAC		0x08
-+#define WM8753_RDAC		0x09
-+#define WM8753_BASS		0x0a
-+#define WM8753_TREBLE		0x0b
-+#define WM8753_ALC1		0x0c
-+#define WM8753_ALC2		0x0d
-+#define WM8753_ALC3		0x0e
-+#define WM8753_NGATE		0x0f
-+#define WM8753_LADC		0x10
-+#define WM8753_RADC		0x11
-+#define WM8753_ADCTL1		0x12
-+#define WM8753_3D		0x13
-+#define WM8753_PWR1		0x14
-+#define WM8753_PWR2		0x15
-+#define WM8753_PWR3		0x16
-+#define WM8753_PWR4		0x17
-+#define WM8753_ID		0x18
-+#define WM8753_INTPOL		0x19
-+#define WM8753_INTEN		0x1a
-+#define WM8753_GPIO1		0x1b
-+#define WM8753_GPIO2		0x1c
-+#define WM8753_RESET		0x1f
-+#define WM8753_RECMIX1		0x20
-+#define WM8753_RECMIX2		0x21
-+#define WM8753_LOUTM1		0x22
-+#define WM8753_LOUTM2		0x23
-+#define WM8753_ROUTM1		0x24
-+#define WM8753_ROUTM2		0x25
-+#define WM8753_MOUTM1		0x26
-+#define WM8753_MOUTM2		0x27
-+#define WM8753_LOUT1V		0x28
-+#define WM8753_ROUT1V		0x29
-+#define WM8753_LOUT2V		0x2a
-+#define WM8753_ROUT2V		0x2b
-+#define WM8753_MOUTV		0x2c
-+#define WM8753_OUTCTL		0x2d
-+#define WM8753_ADCIN		0x2e
-+#define WM8753_INCTL1		0x2f
-+#define WM8753_INCTL2		0x30
-+#define WM8753_LINVOL		0x31
-+#define WM8753_RINVOL		0x32
-+#define WM8753_MICBIAS		0x33
-+#define WM8753_CLOCK		0x34
-+#define WM8753_PLL1CTL1		0x35
-+#define WM8753_PLL1CTL2		0x36
-+#define WM8753_PLL1CTL3		0x37
-+#define WM8753_PLL1CTL4		0x38
-+#define WM8753_PLL2CTL1		0x39
-+#define WM8753_PLL2CTL2		0x3a
-+#define WM8753_PLL2CTL3		0x3b
-+#define WM8753_PLL2CTL4		0x3c
-+#define WM8753_BIASCTL		0x3d
-+#define WM8753_ADCTL2		0x3f
-+
-+struct wm8753_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+#define WM8753_PLL1			0
-+#define WM8753_PLL2			1
-+
-+/* clock inputs */
-+#define WM8753_MCLK		0
-+#define WM8753_PCMCLK		1
-+
-+/* clock divider id's */
-+#define WM8753_PCMDIV		0
-+#define WM8753_BCLKDIV		1
-+#define WM8753_VXCLKDIV		2
-+
-+/* PCM clock dividers */
-+#define WM8753_PCM_DIV_1	(0 << 6)
-+#define WM8753_PCM_DIV_3	(2 << 6)
-+#define WM8753_PCM_DIV_5_5	(3 << 6)
-+#define WM8753_PCM_DIV_2	(4 << 6)
-+#define WM8753_PCM_DIV_4	(5 << 6)
-+#define WM8753_PCM_DIV_6	(6 << 6)
-+#define WM8753_PCM_DIV_8	(7 << 6)
-+
-+/* BCLK clock dividers */
-+#define WM8753_BCLK_DIV_1	(0 << 3)
-+#define WM8753_BCLK_DIV_2	(1 << 3)
-+#define WM8753_BCLK_DIV_4	(2 << 3)
-+#define WM8753_BCLK_DIV_8	(3 << 3)
-+#define WM8753_BCLK_DIV_16	(4 << 3)
-+
-+/* VXCLK clock dividers */
-+#define WM8753_VXCLK_DIV_1	(0 << 6)
-+#define WM8753_VXCLK_DIV_2	(1 << 6)
-+#define WM8753_VXCLK_DIV_4	(2 << 6)
-+#define WM8753_VXCLK_DIV_8	(3 << 6)
-+#define WM8753_VXCLK_DIV_16	(4 << 6)
-+
-+#define WM8753_DAI_HIFI		0
-+#define WM8753_DAI_VOICE		1
-+
-+extern struct snd_soc_codec_dai wm8753_dai[2];
-+extern struct snd_soc_codec_device soc_codec_dev_wm8753;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8772.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8772.c
-@@ -0,0 +1,603 @@
-+/*
-+ * wm8772.c  --  WM8772 ALSA Soc Audio driver
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8772.h"
-+
-+#define AUDIO_NAME "WM8772"
-+#define WM8772_VERSION "0.4"
-+
-+/* codec private data */
-+struct wm8772_priv {
-+	unsigned int adcclk;
-+	unsigned int dacclk;
-+};
-+
-+/*
-+ * wm8772 register cache
-+ * We can't read the WM8772 register space when we
-+ * are using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8772_reg[] = {
-+	0x00ff, 0x00ff, 0x0120, 0x0000,  /*  0 */
-+	0x00ff, 0x00ff, 0x00ff, 0x00ff,  /*  4 */
-+	0x00ff, 0x0000, 0x0080, 0x0040,  /*  8 */
-+	0x0000
-+};
-+
-+/*
-+ * read wm8772 register cache
-+ */
-+static inline unsigned int wm8772_read_reg_cache(struct snd_soc_codec * codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg > WM8772_CACHE_REGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8772 register cache
-+ */
-+static inline void wm8772_write_reg_cache(struct snd_soc_codec * codec,
-+	unsigned int reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg > WM8772_CACHE_REGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+static int wm8772_write(struct snd_soc_codec * codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8772 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8772_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -1;
-+}
-+
-+#define wm8772_reset(c)	wm8772_write(c, WM8772_RESET, 0)
-+
-+/*
-+ * WM8772 Controls
-+ */
-+static const char *wm8772_zero_flag[] = {"All Ch", "Ch 1", "Ch 2", "Ch3"};
-+
-+static const struct soc_enum wm8772_enum[] = {
-+SOC_ENUM_SINGLE(WM8772_DACCTRL, 0, 4, wm8772_zero_flag),
-+};
-+
-+static const struct snd_kcontrol_new wm8772_snd_controls[] = {
-+
-+SOC_SINGLE("Left1 Playback Volume", WM8772_LDAC1VOL, 0, 255, 0),
-+SOC_SINGLE("Left2 Playback Volume", WM8772_LDAC2VOL, 0, 255, 0),
-+SOC_SINGLE("Left3 Playback Volume", WM8772_LDAC3VOL, 0, 255, 0),
-+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC1VOL, 0, 255, 0),
-+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC2VOL, 0, 255, 0),
-+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC3VOL, 0, 255, 0),
-+SOC_SINGLE("Master Playback Volume", WM8772_MDACVOL, 0, 255, 0),
-+
-+SOC_SINGLE("Playback Switch", WM8772_DACCH, 0, 1, 0),
-+SOC_SINGLE("Capture Switch", WM8772_ADCCTRL, 2, 1, 0),
-+
-+SOC_SINGLE("Demp1 Playback Switch", WM8772_DACCTRL, 6, 1, 0),
-+SOC_SINGLE("Demp2 Playback Switch", WM8772_DACCTRL, 7, 1, 0),
-+SOC_SINGLE("Demp3 Playback Switch", WM8772_DACCTRL, 8, 1, 0),
-+
-+SOC_SINGLE("Phase Invert 1 Switch", WM8772_IFACE, 6, 1, 0),
-+SOC_SINGLE("Phase Invert 2 Switch", WM8772_IFACE, 7, 1, 0),
-+SOC_SINGLE("Phase Invert 3 Switch", WM8772_IFACE, 8, 1, 0),
-+
-+SOC_SINGLE("Playback ZC Switch", WM8772_DACCTRL, 0, 1, 0),
-+
-+SOC_SINGLE("Capture High Pass Switch", WM8772_ADCCTRL, 3, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8772_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8772_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8772_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+	return 0;
-+}
-+
-+static int wm8772_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
-+		int clk_id, unsigned int freq, int dir)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	struct wm8772_priv *wm8772 = codec->private_data;
-+
-+	switch (freq) {
-+	case 4096000:
-+	case 5644800:
-+	case 6144000:
-+	case 8192000:
-+	case 8467000:
-+	case 9216000:
-+	case 11289600:
-+	case 12000000:
-+	case 12288000:
-+	case 16934400:
-+	case 18432000:
-+	case 22579200:
-+	case 24576000:
-+	case 33868800:
-+	case 36864000:
-+		if (clk_id == WM8772_DACCLK) {
-+			wm8772->dacclk = freq;
-+			return 0;
-+		} else if (clk_id == WM8772_ADCCLK) {
-+			wm8772->adcclk = freq;
-+			return 0;
-+		}
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8772_set_dac_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0x1f0;
-+	u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0x1ef;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		diface_ctrl |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		diface |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		diface |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		diface |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		diface |= 0x0007;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		diface |= 0x0008;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8772_write(codec, WM8772_DACRATE, diface_ctrl);
-+	wm8772_write(codec, WM8772_IFACE, diface);
-+	return 0;
-+}
-+
-+static int wm8772_set_adc_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 aiface = 0;
-+	u16 aiface_ctrl = wm8772_read_reg_cache(codec, WM8772_ADCCTRL) & 0x1cf;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		aiface |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		aiface |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		aiface |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		aiface |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		aiface |= 0x0003;
-+		aiface_ctrl |= 0x0010;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		aiface_ctrl |= 0x0020;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8772_write(codec, WM8772_ADCCTRL, aiface_ctrl);
-+	wm8772_write(codec, WM8772_ADCRATE, aiface);
-+	return 0;
-+}
-+
-+static int wm8772_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm8772_priv *wm8772 = codec->private_data;
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+
-+		u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0x1cf;
-+		u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0x3f;
-+
-+		/* bit size */
-+		switch (params_format(params)) {
-+		case SNDRV_PCM_FORMAT_S16_LE:
-+			break;
-+		case SNDRV_PCM_FORMAT_S20_3LE:
-+			diface |= 0x0010;
-+			break;
-+		case SNDRV_PCM_FORMAT_S24_3LE:
-+			diface |= 0x0020;
-+			break;
-+		case SNDRV_PCM_FORMAT_S32_LE:
-+			diface |= 0x0030;
-+			break;
-+		}
-+
-+		/* set rate */
-+		switch (wm8772->dacclk / params_rate(params)) {
-+		case 768:
-+			diface_ctrl |= (0x5 << 6);
-+			break;
-+		case 512:
-+			diface_ctrl |= (0x4 << 6);
-+			break;
-+		case 384:
-+			diface_ctrl |= (0x3 << 6);
-+			break;
-+		case 256:
-+			diface_ctrl |= (0x2 << 6);
-+			break;
-+		case 192:
-+			diface_ctrl |= (0x1 << 6);
-+			break;
-+		}
-+
-+		wm8772_write(codec, WM8772_DACRATE, diface_ctrl);
-+		wm8772_write(codec, WM8772_IFACE, diface);
-+
-+	} else {
-+
-+		u16 aiface = wm8772_read_reg_cache(codec, WM8772_ADCRATE) & 0x113;
-+
-+		/* bit size */
-+		switch (params_format(params)) {
-+		case SNDRV_PCM_FORMAT_S16_LE:
-+			break;
-+		case SNDRV_PCM_FORMAT_S20_3LE:
-+			aiface |= 0x0004;
-+			break;
-+		case SNDRV_PCM_FORMAT_S24_LE:
-+			aiface |= 0x0008;
-+			break;
-+		case SNDRV_PCM_FORMAT_S32_LE:
-+			aiface |= 0x000c;
-+			break;
-+		}
-+
-+		/* set rate */
-+		switch (wm8772->adcclk / params_rate(params)) {
-+		case 768:
-+			aiface |= (0x5 << 5);
-+			break;
-+		case 512:
-+			aiface |= (0x4 << 5);
-+			break;
-+		case 384:
-+			aiface |= (0x3 << 5);
-+			break;
-+		case 256:
-+			aiface |= (0x2 << 5);
-+			break;
-+		}
-+
-+		wm8772_write(codec, WM8772_ADCRATE, aiface);
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8772_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 master = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xffe0;
-+
-+	switch (event) {
-+		case SNDRV_CTL_POWER_D0: /* full On */
-+			/* vref/mid, clk and osc on, dac unmute, active */
-+			wm8772_write(codec, WM8772_DACRATE, master);
-+			break;
-+		case SNDRV_CTL_POWER_D1: /* partial On */
-+		case SNDRV_CTL_POWER_D2: /* partial On */
-+			break;
-+		case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+			/* everything off except vref/vmid, dac mute, inactive */
-+			wm8772_write(codec, WM8772_DACRATE, master | 0x0f);
-+			break;
-+		case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+			/* everything off, dac mute, inactive */
-+			wm8772_write(codec, WM8772_DACRATE, master | 0x1f);
-+			break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+struct snd_soc_codec_dai wm8772_dai[] = {
-+{
-+	.name = "WM8772",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 2,
-+		.channels_max = 6,
-+	},
-+	.ops = {
-+		.hw_params = wm8772_hw_params,
-+	},
-+	.dai_ops = {
-+		.set_fmt = wm8772_set_dac_dai_fmt,
-+		.set_sysclk = wm8772_set_dai_sysclk,
-+	},
-+},
-+{
-+	.name = "WM8772",
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 2,
-+		.channels_max = 2,
-+	},
-+	.ops = {
-+		.hw_params = wm8772_hw_params,
-+	},
-+	.dai_ops = {
-+		.set_fmt = wm8772_set_adc_dai_fmt,
-+		.set_sysclk = wm8772_set_dai_sysclk,
-+	},
-+},
-+};
-+EXPORT_SYMBOL_GPL(wm8772_dai);
-+
-+static int wm8772_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8772_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8772_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8772_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8772 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8772_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "WM8772";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8772_read_reg_cache;
-+	codec->write = wm8772_write;
-+	codec->dapm_event = wm8772_dapm_event;
-+	codec->dai = wm8772_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8772_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8772_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8772_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8772_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8772_reg);
-+
-+	wm8772_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "wm8772: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* set the update bits */
-+	reg = wm8772_read_reg_cache(codec, WM8772_MDACVOL);
-+	wm8772_write(codec, WM8772_MDACVOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_LDAC1VOL);
-+	wm8772_write(codec, WM8772_LDAC1VOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_LDAC2VOL);
-+	wm8772_write(codec, WM8772_LDAC2VOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_LDAC3VOL);
-+	wm8772_write(codec, WM8772_LDAC3VOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_RDAC1VOL);
-+	wm8772_write(codec, WM8772_RDAC1VOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_RDAC2VOL);
-+	wm8772_write(codec, WM8772_RDAC2VOL, reg | 0x0100);
-+	reg = wm8772_read_reg_cache(codec, WM8772_RDAC3VOL);
-+	wm8772_write(codec, WM8772_RDAC3VOL, reg | 0x0100);
-+
-+	wm8772_add_controls(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8772: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8772_socdev;
-+
-+static int wm8772_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8772_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	struct wm8772_priv *wm8772;
-+	int ret = 0;
-+
-+	printk(KERN_INFO "WM8772 Audio Codec %s", WM8772_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	wm8772 = kzalloc(sizeof(struct wm8772_priv), GFP_KERNEL);
-+	if (wm8772 == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+
-+	codec->private_data = wm8772;
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8772_socdev = socdev;
-+
-+	/* Add other interfaces here */
-+#warning do SPI device probe here and then call wm8772_init()
-+
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8772_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	kfree(codec->private_data);
-+	kfree(codec->reg_cache);
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8772 = {
-+	.probe = 	wm8772_probe,
-+	.remove = 	wm8772_remove,
-+	.suspend = 	wm8772_suspend,
-+	.resume =	wm8772_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8772);
-+
-+MODULE_DESCRIPTION("ASoC WM8772 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8772.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8772.h
-@@ -0,0 +1,46 @@
-+/*
-+ * wm8772.h  --  audio driver for WM8772
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ */
-+
-+#ifndef _WM8772_H
-+#define _WM8772_H
-+
-+/* WM8772 register space */
-+
-+#define WM8772_LDAC1VOL   0x00
-+#define WM8772_RDAC1VOL   0x01
-+#define WM8772_DACCH      0x02
-+#define WM8772_IFACE      0x03
-+#define WM8772_LDAC2VOL   0x04
-+#define WM8772_RDAC2VOL   0x05
-+#define WM8772_LDAC3VOL   0x06
-+#define WM8772_RDAC3VOL   0x07
-+#define WM8772_MDACVOL    0x08
-+#define WM8772_DACCTRL    0x09
-+#define WM8772_DACRATE    0x0a
-+#define WM8772_ADCRATE    0x0b
-+#define WM8772_ADCCTRL    0x0c
-+#define WM8772_RESET	  0x1f
-+
-+#define WM8772_CACHE_REGNUM 	10
-+
-+#define WM8772_DACCLK	0
-+#define WM8772_ADCCLK	1
-+
-+#define WM8753_DAI_DAC		0
-+#define WM8753_DAI_ADC		1
-+
-+extern struct snd_soc_codec_dai wm8772_dai[2];
-+extern struct snd_soc_codec_device soc_codec_dev_wm8772;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8971.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8971.c
-@@ -0,0 +1,971 @@
-+/*
-+ * wm8971.c  --  WM8971 ALSA SoC Audio driver
-+ *
-+ * Copyright 2005 Lab126, Inc.
-+ *
-+ * Author: Kenneth Kiraly <kiraly at lab126.com>
-+ *
-+ * Based on wm8753.c by Liam Girdwood
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8971.h"
-+
-+#define AUDIO_NAME "wm8971"
-+#define WM8971_VERSION "0.9"
-+
-+#undef	WM8971_DEBUG
-+
-+#ifdef WM8971_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+#define	WM8971_REG_COUNT		43
-+
-+static struct workqueue_struct *wm8971_workq = NULL;
-+
-+/* codec private data */
-+struct wm8971_priv {
-+	unsigned int sysclk;
-+};
-+
-+/*
-+ * wm8971 register cache
-+ * We can't read the WM8971 register space when we
-+ * are using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8971_reg[] = {
-+	0x0097, 0x0097, 0x0079, 0x0079,  /*  0 */
-+	0x0000, 0x0008, 0x0000, 0x000a,  /*  4 */
-+	0x0000, 0x0000, 0x00ff, 0x00ff,  /*  8 */
-+	0x000f, 0x000f, 0x0000, 0x0000,  /* 12 */
-+	0x0000, 0x007b, 0x0000, 0x0032,  /* 16 */
-+	0x0000, 0x00c3, 0x00c3, 0x00c0,  /* 20 */
-+	0x0000, 0x0000, 0x0000, 0x0000,  /* 24 */
-+	0x0000, 0x0000, 0x0000, 0x0000,  /* 28 */
-+	0x0000, 0x0000, 0x0050, 0x0050,  /* 32 */
-+	0x0050, 0x0050, 0x0050, 0x0050,  /* 36 */
-+	0x0079, 0x0079, 0x0079,          /* 40 */
-+};
-+
-+static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg < WM8971_REG_COUNT)
-+		return cache[reg];
-+
-+	return -1;
-+}
-+
-+static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg < WM8971_REG_COUNT)
-+		cache[reg] = value;
-+}
-+
-+static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8753 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8971_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8971_reset(c)	wm8971_write(c, WM8971_RESET, 0)
-+
-+/* WM8971 Controls */
-+static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" };
-+static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz",
-+	"200Hz @ 48kHz" };
-+static const char *wm8971_treble[] = { "8kHz", "4kHz" };
-+static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" };
-+static const char *wm8971_ng_type[] = { "Constant PGA Gain",
-+	"Mute ADC Output" };
-+static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
-+static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)",
-+	"Mono (Right)", "Digital Mono"};
-+static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" };
-+static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA",
-+	"Differential"};
-+static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA",
-+	"Differential"};
-+static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"};
-+static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"};
-+static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert",
-+	"L + R Invert"};
-+
-+static const struct soc_enum wm8971_enum[] = {
-+	SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass),			/* 0 */
-+	SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter),
-+	SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble),
-+	SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func),
-+	SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type),        /* 4 */
-+	SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp),
-+	SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux),
-+	SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase),
-+	SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux),     /* 8 */
-+	SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux),
-+	SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel),
-+	SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel),
-+	SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol),        /* 12 */
-+	SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux),
-+};
-+
-+static const struct snd_kcontrol_new wm8971_snd_controls[] = {
-+	SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0),
-+	SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, 6, 1, 0),
-+	SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1),
-+
-+	SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V,
-+		WM8971_ROUT1V, 7, 1, 0),
-+	SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V,
-+		WM8971_ROUT2V, 7, 1, 0),
-+	SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0),
-+
-+	SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0),
-+
-+	SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1,
-+		WM8971_LOUTM2, 4, 7, 1),
-+	SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1,
-+		WM8971_ROUTM2, 4, 7, 1),
-+	SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1,
-+		WM8971_MOUTM2, 4, 7, 1),
-+
-+	SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V,
-+		WM8971_ROUT1V, 0, 127, 0),
-+	SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V,
-+		WM8971_ROUT2V, 0, 127, 0),
-+
-+	SOC_ENUM("Bass Boost", wm8971_enum[0]),
-+	SOC_ENUM("Bass Filter", wm8971_enum[1]),
-+	SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1),
-+
-+	SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0),
-+	SOC_ENUM("Treble Cut-off", wm8971_enum[2]),
-+
-+	SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1),
-+
-+	SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0),
-+	SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0),
-+
-+	SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0),
-+	SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0),
-+	SOC_ENUM("ALC Capture Function", wm8971_enum[3]),
-+	SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0),
-+	SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0),
-+	SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0),
-+	SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0),
-+	SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0),
-+	SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]),
-+	SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0),
-+
-+	SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0),
-+	SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0),
-+
-+    SOC_ENUM("Playback De-emphasis", wm8971_enum[5]),
-+	SOC_ENUM("Playback Function", wm8971_enum[6]),
-+	SOC_ENUM("Playback Phase", wm8971_enum[7]),
-+
-+	SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
-+};
-+
-+/* add non-DAPM controls */
-+static int wm8971_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8971_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+	return 0;
-+}
-+
-+/*
-+ * DAPM Controls
-+ */
-+
-+/* Left Mixer */
-+static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0),
-+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0),
-+};
-+
-+/* Right Mixer */
-+static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0),
-+SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0),
-+};
-+
-+/* Mono Mixer */
-+static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0),
-+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0),
-+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0),
-+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0),
-+};
-+
-+/* Left Line Mux */
-+static const struct snd_kcontrol_new wm8971_left_line_controls =
-+SOC_DAPM_ENUM("Route", wm8971_enum[8]);
-+
-+/* Right Line Mux */
-+static const struct snd_kcontrol_new wm8971_right_line_controls =
-+SOC_DAPM_ENUM("Route", wm8971_enum[9]);
-+
-+/* Left PGA Mux */
-+static const struct snd_kcontrol_new wm8971_left_pga_controls =
-+SOC_DAPM_ENUM("Route", wm8971_enum[10]);
-+
-+/* Right PGA Mux */
-+static const struct snd_kcontrol_new wm8971_right_pga_controls =
-+SOC_DAPM_ENUM("Route", wm8971_enum[11]);
-+
-+/* Mono ADC Mux */
-+static const struct snd_kcontrol_new wm8971_monomux_controls =
-+SOC_DAPM_ENUM("Route", wm8971_enum[13]);
-+
-+static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = {
-+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
-+		&wm8971_left_mixer_controls[0],
-+		ARRAY_SIZE(wm8971_left_mixer_controls)),
-+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
-+		&wm8971_right_mixer_controls[0],
-+		ARRAY_SIZE(wm8971_right_mixer_controls)),
-+	SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0,
-+		&wm8971_mono_mixer_controls[0],
-+		ARRAY_SIZE(wm8971_mono_mixer_controls)),
-+
-+	SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0),
-+	SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0),
-+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0),
-+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0),
-+	SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0),
-+
-+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0),
-+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0),
-+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0),
-+
-+	SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0,
-+		&wm8971_left_pga_controls),
-+	SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0,
-+		&wm8971_right_pga_controls),
-+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
-+		&wm8971_left_line_controls),
-+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
-+		&wm8971_right_line_controls),
-+
-+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
-+		&wm8971_monomux_controls),
-+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
-+		&wm8971_monomux_controls),
-+
-+	SND_SOC_DAPM_OUTPUT("LOUT1"),
-+	SND_SOC_DAPM_OUTPUT("ROUT1"),
-+	SND_SOC_DAPM_OUTPUT("LOUT2"),
-+	SND_SOC_DAPM_OUTPUT("ROUT2"),
-+	SND_SOC_DAPM_OUTPUT("MONO"),
-+
-+	SND_SOC_DAPM_INPUT("LINPUT1"),
-+	SND_SOC_DAPM_INPUT("RINPUT1"),
-+	SND_SOC_DAPM_INPUT("MIC"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* left mixer */
-+	{"Left Mixer", "Playback Switch", "Left DAC"},
-+	{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
-+	{"Left Mixer", "Right Playback Switch", "Right DAC"},
-+	{"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
-+
-+	/* right mixer */
-+	{"Right Mixer", "Left Playback Switch", "Left DAC"},
-+	{"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
-+	{"Right Mixer", "Playback Switch", "Right DAC"},
-+	{"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
-+
-+	/* left out 1 */
-+	{"Left Out 1", NULL, "Left Mixer"},
-+	{"LOUT1", NULL, "Left Out 1"},
-+
-+	/* left out 2 */
-+	{"Left Out 2", NULL, "Left Mixer"},
-+	{"LOUT2", NULL, "Left Out 2"},
-+
-+	/* right out 1 */
-+	{"Right Out 1", NULL, "Right Mixer"},
-+	{"ROUT1", NULL, "Right Out 1"},
-+
-+	/* right out 2 */
-+	{"Right Out 2", NULL, "Right Mixer"},
-+	{"ROUT2", NULL, "Right Out 2"},
-+
-+	/* mono mixer */
-+	{"Mono Mixer", "Left Playback Switch", "Left DAC"},
-+	{"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
-+	{"Mono Mixer", "Right Playback Switch", "Right DAC"},
-+	{"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
-+
-+	/* mono out */
-+	{"Mono Out", NULL, "Mono Mixer"},
-+	{"MONO1", NULL, "Mono Out"},
-+
-+	/* Left Line Mux */
-+	{"Left Line Mux", "Line", "LINPUT1"},
-+	{"Left Line Mux", "PGA", "Left PGA Mux"},
-+	{"Left Line Mux", "Differential", "Differential Mux"},
-+
-+	/* Right Line Mux */
-+	{"Right Line Mux", "Line", "RINPUT1"},
-+	{"Right Line Mux", "Mic", "MIC"},
-+	{"Right Line Mux", "PGA", "Right PGA Mux"},
-+	{"Right Line Mux", "Differential", "Differential Mux"},
-+
-+	/* Left PGA Mux */
-+	{"Left PGA Mux", "Line", "LINPUT1"},
-+	{"Left PGA Mux", "Differential", "Differential Mux"},
-+
-+	/* Right PGA Mux */
-+	{"Right PGA Mux", "Line", "RINPUT1"},
-+	{"Right PGA Mux", "Differential", "Differential Mux"},
-+
-+	/* Differential Mux */
-+	{"Differential Mux", "Line", "LINPUT1"},
-+	{"Differential Mux", "Line", "RINPUT1"},
-+
-+	/* Left ADC Mux */
-+	{"Left ADC Mux", "Stereo", "Left PGA Mux"},
-+	{"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
-+	{"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
-+
-+	/* Right ADC Mux */
-+	{"Right ADC Mux", "Stereo", "Right PGA Mux"},
-+	{"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
-+	{"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
-+
-+	/* ADC */
-+	{"Left ADC", NULL, "Left ADC Mux"},
-+	{"Right ADC", NULL, "Right ADC Mux"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8971_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8971_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8971_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct _coeff_div {
-+	u32 mclk;
-+	u32 rate;
-+	u16 fs;
-+	u8 sr:5;
-+	u8 usb:1;
-+};
-+
-+/* codec hifi mclk clock divider coefficients */
-+static const struct _coeff_div coeff_div[] = {
-+	/* 8k */
-+	{12288000, 8000, 1536, 0x6, 0x0},
-+	{11289600, 8000, 1408, 0x16, 0x0},
-+	{18432000, 8000, 2304, 0x7, 0x0},
-+	{16934400, 8000, 2112, 0x17, 0x0},
-+	{12000000, 8000, 1500, 0x6, 0x1},
-+
-+	/* 11.025k */
-+	{11289600, 11025, 1024, 0x18, 0x0},
-+	{16934400, 11025, 1536, 0x19, 0x0},
-+	{12000000, 11025, 1088, 0x19, 0x1},
-+
-+	/* 16k */
-+	{12288000, 16000, 768, 0xa, 0x0},
-+	{18432000, 16000, 1152, 0xb, 0x0},
-+	{12000000, 16000, 750, 0xa, 0x1},
-+
-+	/* 22.05k */
-+	{11289600, 22050, 512, 0x1a, 0x0},
-+	{16934400, 22050, 768, 0x1b, 0x0},
-+	{12000000, 22050, 544, 0x1b, 0x1},
-+
-+	/* 32k */
-+	{12288000, 32000, 384, 0xc, 0x0},
-+	{18432000, 32000, 576, 0xd, 0x0},
-+	{12000000, 32000, 375, 0xa, 0x1},
-+
-+	/* 44.1k */
-+	{11289600, 44100, 256, 0x10, 0x0},
-+	{16934400, 44100, 384, 0x11, 0x0},
-+	{12000000, 44100, 272, 0x11, 0x1},
-+
-+	/* 48k */
-+	{12288000, 48000, 256, 0x0, 0x0},
-+	{18432000, 48000, 384, 0x1, 0x0},
-+	{12000000, 48000, 250, 0x0, 0x1},
-+
-+	/* 88.2k */
-+	{11289600, 88200, 128, 0x1e, 0x0},
-+	{16934400, 88200, 192, 0x1f, 0x0},
-+	{12000000, 88200, 136, 0x1f, 0x1},
-+
-+	/* 96k */
-+	{12288000, 96000, 128, 0xe, 0x0},
-+	{18432000, 96000, 192, 0xf, 0x0},
-+	{12000000, 96000, 125, 0xe, 0x1},
-+};
-+
-+static int get_coeff(int mclk, int rate)
-+{
-+	int i;
-+
-+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
-+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
-+			return i;
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8971_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
-+		int clk_id, unsigned int freq, int dir)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	struct wm8971_priv *wm8971 = codec->private_data;
-+
-+	switch (freq) {
-+	case 11289600:
-+	case 12000000:
-+	case 12288000:
-+	case 16934400:
-+	case 18432000:
-+		wm8971->sysclk = freq;
-+		return 0;
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8971_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = 0;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		iface = 0x0040;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		iface |= 0x0013;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0090;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0080;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0010;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8971_write(codec, WM8971_IFACE, iface);
-+	return 0;
-+}
-+
-+static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm8971_priv *wm8971 = codec->private_data;
-+	u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
-+	u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
-+	int coeff = get_coeff(wm8971->sysclk, params_rate(params));
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0008;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		iface |= 0x000c;
-+		break;
-+	}
-+
-+	/* set iface & srate */
-+	wm8971_write(codec, WM8971_IFACE, iface);
-+	if (coeff >= 0)
-+		wm8971_write(codec, WM8971_SRATE, srate |
-+			(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
-+
-+	return 0;
-+}
-+
-+static int wm8971_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7;
-+
-+	if (mute)
-+		wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8);
-+	else
-+		wm8971_write(codec, WM8971_ADCDAC, mute_reg);
-+	return 0;
-+}
-+
-+static int wm8971_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* set vmid to 50k and unmute dac */
-+		wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		/* set vmid to 5k for quick power up */
-+		wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x01c0);
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* mute dac and set vmid to 500k, enable VREF */
-+		wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		wm8971_write(codec, WM8971_PWR1, 0x0001);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+#define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8971_dai = {
-+	.name = "WM8971",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8971_RATES,
-+		.formats = WM8971_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8971_RATES,
-+		.formats = WM8971_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8971_pcm_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8971_mute,
-+		.set_fmt = wm8971_set_dai_fmt,
-+		.set_sysclk = wm8971_set_dai_sysclk,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8971_dai);
-+
-+static void wm8971_work(struct work_struct *work)
-+{
-+	struct snd_soc_codec *codec =
-+		container_of(work, struct snd_soc_codec, delayed_work.work);
-+	wm8971_dapm_event(codec, codec->dapm_state);
-+}
-+
-+static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8971_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) {
-+		if (i + 1 == WM8971_RESET)
-+			continue;
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+
-+	wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* charge wm8971 caps */
-+	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-+		wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2);
-+		codec->dapm_state = SNDRV_CTL_POWER_D0;
-+		queue_delayed_work(wm8971_workq, &codec->delayed_work,
-+			msecs_to_jiffies(1000));
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8971_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "WM8971";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8971_read_reg_cache;
-+	codec->write = wm8971_write;
-+	codec->dapm_event = wm8971_dapm_event;
-+	codec->dai = &wm8971_dai;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8971_reg);
-+	codec->num_dai = 1;
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8971_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8971_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8971_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8971_reg);
-+
-+	wm8971_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "wm8971: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* charge output caps */
-+	wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2);
-+	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
-+	queue_delayed_work(wm8971_workq, &codec->delayed_work,
-+		msecs_to_jiffies(1000));
-+
-+	/* set the update bits */
-+	reg = wm8971_read_reg_cache(codec, WM8971_LDAC);
-+	wm8971_write(codec, WM8971_LDAC, reg | 0x0100);
-+	reg = wm8971_read_reg_cache(codec, WM8971_RDAC);
-+	wm8971_write(codec, WM8971_RDAC, reg | 0x0100);
-+
-+	reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V);
-+	wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100);
-+	reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V);
-+	wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100);
-+
-+	reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V);
-+	wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100);
-+	reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V);
-+	wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100);
-+
-+	reg = wm8971_read_reg_cache(codec, WM8971_LINVOL);
-+	wm8971_write(codec, WM8971_LINVOL, reg | 0x0100);
-+	reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
-+	wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
-+
-+	wm8971_add_controls(codec);
-+	wm8971_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8971: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+static struct snd_soc_device *wm8971_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8731 2 wire address is determined by GPIO5
-+ * state during powerup.
-+ *    low  = 0x1a
-+ *    high = 0x1b
-+ */
-+#define I2C_DRIVERID_WM8971 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8971_i2c_driver;
-+static struct i2c_client client_template;
-+
-+static int wm8971_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8971_socdev;
-+	struct wm8971_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+
-+	i2c_set_clientdata(i2c, codec);
-+
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8971_init(socdev);
-+	if (ret < 0) {
-+		err("failed to initialise WM8971\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int wm8971_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8971_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8971_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8971_i2c_driver = {
-+	.driver = {
-+		.name = "WM8971 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8971,
-+	.attach_adapter = wm8971_i2c_attach,
-+	.detach_client =  wm8971_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8971",
-+	.driver = &wm8971_i2c_driver,
-+};
-+#endif
-+
-+static int wm8971_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8971_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	struct wm8971_priv *wm8971;
-+	int ret = 0;
-+
-+	info("WM8971 Audio Codec %s", WM8971_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	wm8971 = kzalloc(sizeof(struct wm8971_priv), GFP_KERNEL);
-+	if (wm8971 == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+
-+	codec->private_data = wm8971;
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+	wm8971_socdev = socdev;
-+
-+	INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
-+	wm8971_workq = create_workqueue("wm8971");
-+	if (wm8971_workq == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8971_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+		/* Add other interfaces here */
-+#endif
-+
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8971_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	if (wm8971_workq)
-+		destroy_workqueue(wm8971_workq);
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8971_i2c_driver);
-+#endif
-+	kfree(codec->private_data);
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8971 = {
-+	.probe = 	wm8971_probe,
-+	.remove = 	wm8971_remove,
-+	.suspend = 	wm8971_suspend,
-+	.resume =	wm8971_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
-+
-+MODULE_DESCRIPTION("ASoC WM8971 driver");
-+MODULE_AUTHOR("Lab126");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8971.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8971.h
-@@ -0,0 +1,63 @@
-+/*
-+ * wm8971.h  --  audio driver for WM8971
-+ *
-+ * Copyright 2005 Lab126, Inc.
-+ *
-+ * Author: Kenneth Kiraly <kiraly at lab126.com>
-+ *
-+ * This program is free software; you can redistribute  it and/or modify it
-+ * under  the terms of  the GNU General  Public License as published by the
-+ * Free Software Foundation;  either version 2 of the  License, or (at your
-+ * option) any later version.
-+ *
-+ */
-+
-+#ifndef _WM8971_H
-+#define _WM8971_H
-+
-+#define WM8971_LINVOL	0x00
-+#define WM8971_RINVOL	0x01
-+#define WM8971_LOUT1V	0x02
-+#define WM8971_ROUT1V	0x03
-+#define WM8971_ADCDAC	0x05
-+#define WM8971_IFACE	0x07
-+#define WM8971_SRATE	0x08
-+#define WM8971_LDAC		0x0a
-+#define WM8971_RDAC		0x0b
-+#define WM8971_BASS		0x0c
-+#define WM8971_TREBLE	0x0d
-+#define WM8971_RESET	0x0f
-+#define WM8971_ALC1		0x11
-+#define	WM8971_ALC2		0x12
-+#define	WM8971_ALC3		0x13
-+#define WM8971_NGATE	0x14
-+#define WM8971_LADC		0x15
-+#define WM8971_RADC		0x16
-+#define	WM8971_ADCTL1	0x17
-+#define	WM8971_ADCTL2	0x18
-+#define WM8971_PWR1		0x19
-+#define WM8971_PWR2		0x1a
-+#define	WM8971_ADCTL3	0x1b
-+#define WM8971_ADCIN	0x1f
-+#define	WM8971_LADCIN	0x20
-+#define	WM8971_RADCIN	0x21
-+#define WM8971_LOUTM1	0x22
-+#define WM8971_LOUTM2	0x23
-+#define WM8971_ROUTM1	0x24
-+#define WM8971_ROUTM2	0x25
-+#define WM8971_MOUTM1	0x26
-+#define WM8971_MOUTM2	0x27
-+#define WM8971_LOUT2V	0x28
-+#define WM8971_ROUT2V	0x29
-+#define WM8971_MOUTV	0x2A
-+
-+#define WM8971_SYSCLK	0
-+
-+struct wm8971_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8971_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8971;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8974.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8974.c
-@@ -0,0 +1,873 @@
-+/*
-+ * wm8974.c  --  WM8974 ALSA Soc Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ *
-+ * Author: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8974.h"
-+
-+#define AUDIO_NAME "wm8974"
-+#define WM8974_VERSION "0.6"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8974_DEBUG 0
-+
-+#ifdef WM8974_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8974;
-+
-+/*
-+ * wm8974 register cache
-+ * We can't read the WM8974 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0050, 0x0000, 0x0140, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x00ff,
-+    0x0000, 0x0000, 0x0100, 0x00ff,
-+    0x0000, 0x0000, 0x012c, 0x002c,
-+    0x002c, 0x002c, 0x002c, 0x0000,
-+    0x0032, 0x0000, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0038, 0x000b, 0x0032, 0x0000,
-+    0x0008, 0x000c, 0x0093, 0x00e9,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0003, 0x0010, 0x0000, 0x0000,
-+    0x0000, 0x0002, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0039, 0x0000,
-+    0x0000,
-+};
-+
-+/*
-+ * read wm8974 register cache
-+ */
-+static inline unsigned int wm8974_read_reg_cache(struct snd_soc_codec * codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8974_RESET)
-+		return 0;
-+	if (reg >= WM8974_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8974 register cache
-+ */
-+static inline void wm8974_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8974_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8974 register space
-+ */
-+static int wm8974_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8974 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8974_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8974_reset(c)	wm8974_write(c, WM8974_RESET, 0)
-+
-+static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" };
-+static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
-+static const char *wm8974_eqmode[] = {"Capture", "Playback" };
-+static const char *wm8974_bw[] = {"Narrow", "Wide" };
-+static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
-+static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
-+static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
-+static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
-+static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
-+static const char *wm8974_alc[] = {"ALC", "Limiter" };
-+
-+static const struct soc_enum wm8974_enum[] = {
-+	SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */
-+	SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */
-+	SOC_ENUM_SINGLE(WM8974_DAC,  4, 4, wm8974_deemp),
-+	SOC_ENUM_SINGLE(WM8974_EQ1,  8, 2, wm8974_eqmode),
-+
-+	SOC_ENUM_SINGLE(WM8974_EQ1,  5, 4, wm8974_eq1),
-+	SOC_ENUM_SINGLE(WM8974_EQ2,  8, 2, wm8974_bw),
-+	SOC_ENUM_SINGLE(WM8974_EQ2,  5, 4, wm8974_eq2),
-+	SOC_ENUM_SINGLE(WM8974_EQ3,  8, 2, wm8974_bw),
-+
-+	SOC_ENUM_SINGLE(WM8974_EQ3,  5, 4, wm8974_eq3),
-+	SOC_ENUM_SINGLE(WM8974_EQ4,  8, 2, wm8974_bw),
-+	SOC_ENUM_SINGLE(WM8974_EQ4,  5, 4, wm8974_eq4),
-+	SOC_ENUM_SINGLE(WM8974_EQ5,  8, 2, wm8974_bw),
-+
-+	SOC_ENUM_SINGLE(WM8974_EQ5,  5, 4, wm8974_eq5),
-+	SOC_ENUM_SINGLE(WM8974_ALC3,  8, 2, wm8974_alc),
-+};
-+
-+static const struct snd_kcontrol_new wm8974_snd_controls[] = {
-+
-+SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0),
-+
-+SOC_ENUM("DAC Companding", wm8974_enum[1]),
-+SOC_ENUM("ADC Companding", wm8974_enum[0]),
-+
-+SOC_ENUM("Playback De-emphasis", wm8974_enum[2]),
-+SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0),
-+
-+SOC_SINGLE("PCM Volume", WM8974_DACVOL, 0, 127, 0),
-+
-+SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0),
-+SOC_SINGLE("ADC Inversion Switch", WM8974_COMP, 0, 1, 0),
-+
-+SOC_SINGLE("Capture Volume", WM8974_ADCVOL,  0, 127, 0),
-+
-+SOC_ENUM("Equaliser Function", wm8974_enum[3]),
-+SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]),
-+SOC_SINGLE("EQ1 Volume", WM8974_EQ1,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]),
-+SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]),
-+SOC_SINGLE("EQ2 Volume", WM8974_EQ2,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]),
-+SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]),
-+SOC_SINGLE("EQ3 Volume", WM8974_EQ3,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]),
-+SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]),
-+SOC_SINGLE("EQ4 Volume", WM8974_EQ4,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]),
-+SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]),
-+SOC_SINGLE("EQ5 Volume", WM8974_EQ5,  0, 31, 1),
-+
-+SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1,  8, 1, 0),
-+SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1,  4, 15, 0),
-+SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1,  0, 15, 0),
-+
-+SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2,  4, 7, 0),
-+SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Enable Switch", WM8974_ALC1,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1,  3, 7, 0),
-+SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1,  0, 7, 0),
-+
-+SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Hold", WM8974_ALC2,  4, 7, 0),
-+SOC_SINGLE("ALC Capture Target", WM8974_ALC2,  0, 15, 0),
-+
-+SOC_ENUM("ALC Capture Mode", wm8974_enum[13]),
-+SOC_SINGLE("ALC Capture Decay", WM8974_ALC3,  4, 15, 0),
-+SOC_SINGLE("ALC Capture Attack", WM8974_ALC3,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE,  3, 1, 0),
-+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE,  0, 7, 0),
-+
-+SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA,  7, 1, 0),
-+SOC_SINGLE("Capture PGA Volume", WM8974_INPPGA,  0, 63, 0),
-+
-+SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL,  7, 1, 0),
-+SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL,  6, 1, 1),
-+SOC_SINGLE("Speaker Playback Volume", WM8974_SPKVOL,  0, 63, 0),
-+
-+SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST,  8, 1, 0),
-+SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8974_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8974_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8974_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Speaker Output Mixer */
-+static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1),
-+};
-+
-+/* Mono Output Mixer */
-+static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 1),
-+};
-+
-+/* AUX Input boost vol */
-+static const struct snd_kcontrol_new wm8974_aux_boost_controls =
-+SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
-+
-+/* Mic Input boost vol */
-+static const struct snd_kcontrol_new wm8974_mic_boost_controls =
-+SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
-+
-+/* Capture boost switch */
-+static const struct snd_kcontrol_new wm8974_capture_boost_controls =
-+SOC_DAPM_SINGLE("Capture Boost Switch", WM8974_INPPGA,  6, 1, 0);
-+
-+/* Aux In to PGA */
-+static const struct snd_kcontrol_new wm8974_aux_capture_boost_controls =
-+SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8974_INPPGA,  2, 1, 0);
-+
-+/* Mic P In to PGA */
-+static const struct snd_kcontrol_new wm8974_micp_capture_boost_controls =
-+SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8974_INPPGA,  0, 1, 0);
-+
-+/* Mic N In to PGA */
-+static const struct snd_kcontrol_new wm8974_micn_capture_boost_controls =
-+SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8974_INPPGA,  1, 1, 0);
-+
-+static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
-+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
-+	&wm8974_speaker_mixer_controls[0],
-+	ARRAY_SIZE(wm8974_speaker_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0,
-+	&wm8974_mono_mixer_controls[0],
-+	ARRAY_SIZE(wm8974_mono_mixer_controls)),
-+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0),
-+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER3, 0, 0),
-+SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic PGA", WM8974_POWER2, 2, 0, NULL, 0),
-+
-+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8974_aux_boost_controls, 1),
-+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8974_mic_boost_controls, 1),
-+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8974_capture_boost_controls),
-+
-+SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, NULL, 0),
-+
-+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0),
-+
-+SND_SOC_DAPM_INPUT("MICN"),
-+SND_SOC_DAPM_INPUT("MICP"),
-+SND_SOC_DAPM_INPUT("AUX"),
-+SND_SOC_DAPM_OUTPUT("MONOOUT"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* Mono output mixer */
-+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
-+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Speaker output mixer */
-+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
-+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Outputs */
-+	{"Mono Out", NULL, "Mono Mixer"},
-+	{"MONOOUT", NULL, "Mono Out"},
-+	{"SpkN Out", NULL, "Speaker Mixer"},
-+	{"SpkP Out", NULL, "Speaker Mixer"},
-+	{"SPKOUTN", NULL, "SpkN Out"},
-+	{"SPKOUTP", NULL, "SpkP Out"},
-+
-+	/* Boost Mixer */
-+	{"Boost Mixer", NULL, "ADC"},
-+    {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"},
-+	{"Aux Boost", "Aux Volume", "Boost Mixer"},
-+    {"Capture Boost", "Capture Switch", "Boost Mixer"},
-+	{"Mic Boost", "Mic Volume", "Boost Mixer"},
-+
-+	/* Inputs */
-+	{"MICP", NULL, "Mic Boost"},
-+	{"MICN", NULL, "Mic PGA"},
-+	{"Mic PGA", NULL, "Capture Boost"},
-+	{"AUX", NULL, "Aux Input"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8974_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8974_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8974_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct pll_ {
-+	unsigned int in_hz, out_hz;
-+	unsigned int pre:4; /* prescale - 1 */
-+	unsigned int n:4;
-+	unsigned int k;
-+};
-+
-+struct pll_ pll[] = {
-+	{12000000, 11289600, 0, 7, 0x86c220},
-+	{12000000, 12288000, 0, 8, 0x3126e8},
-+	{13000000, 11289600, 0, 6, 0xf28bd4},
-+	{13000000, 12288000, 0, 7, 0x8fd525},
-+	{12288000, 11289600, 0, 7, 0x59999a},
-+	{11289600, 12288000, 0, 8, 0x80dee9},
-+	/* liam - add more entries */
-+};
-+
-+static int wm8974_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	int i;
-+	u16 reg;
-+
-+	if(freq_in == 0 || freq_out == 0) {
-+		reg = wm8974_read_reg_cache(codec, WM8974_POWER1);
-+		wm8974_write(codec, WM8974_POWER1, reg & 0x1df);
-+		return 0;
-+	}
-+
-+	for(i = 0; i < ARRAY_SIZE(pll); i++) {
-+		if (freq_in == pll[i].in_hz && freq_out == pll[i].out_hz) {
-+			wm8974_write(codec, WM8974_PLLN, (pll[i].pre << 4) | pll[i].n);
-+			wm8974_write(codec, WM8974_PLLK1, pll[i].k >> 18);
-+			wm8974_write(codec, WM8974_PLLK1, (pll[i].k >> 9) && 0x1ff);
-+			wm8974_write(codec, WM8974_PLLK1, pll[i].k && 0x1ff);
-+			reg = wm8974_read_reg_cache(codec, WM8974_POWER1);
-+			wm8974_write(codec, WM8974_POWER1, reg | 0x020);
-+			return 0;
-+		}
-+	}
-+	return -EINVAL;
-+}
-+
-+/*
-+ * Configure WM8974 clock dividers.
-+ */
-+static int wm8974_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8974_OPCLKDIV:
-+		reg = wm8974_read_reg_cache(codec, WM8974_GPIO & 0x1cf);
-+		wm8974_write(codec, WM8974_GPIO, reg | div);
-+		break;
-+	case WM8974_MCLKDIV:
-+		reg = wm8974_read_reg_cache(codec, WM8974_CLOCK & 0x1f);
-+		wm8974_write(codec, WM8974_CLOCK, reg | div);
-+		break;
-+	case WM8974_ADCCLK:
-+		reg = wm8974_read_reg_cache(codec, WM8974_ADC & 0x1f7);
-+		wm8974_write(codec, WM8974_ADC, reg | div);
-+		break;
-+	case WM8974_DACCLK:
-+		reg = wm8974_read_reg_cache(codec, WM8974_DAC & 0x1f7);
-+		wm8974_write(codec, WM8974_DAC, reg | div);
-+		break;
-+	case WM8974_BCLKDIV:
-+		reg = wm8974_read_reg_cache(codec, WM8974_CLOCK & 0x1e3);
-+		wm8974_write(codec, WM8974_CLOCK, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8974_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = 0;
-+	u16 clk = wm8974_read_reg_cache(codec, WM8974_CLOCK) & 0x1fe;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		clk |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0008;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x00018;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0180;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0100;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0080;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8974_write(codec, WM8974_IFACE, iface);
-+	wm8974_write(codec, WM8974_CLOCK, clk);
-+	return 0;
-+}
-+
-+static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 iface = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x19f;
-+	u16 adn = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x1f1;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0020;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0040;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		iface |= 0x0060;
-+		break;
-+	}
-+
-+	/* filter coefficient */
-+	switch (params_rate(params)) {
-+	case SNDRV_PCM_RATE_8000:
-+		adn |= 0x5 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_11025:
-+		adn |= 0x4 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_16000:
-+		adn |= 0x3 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_22050:
-+		adn |= 0x2 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_32000:
-+		adn |= 0x1 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_44100:
-+		break;
-+	}
-+
-+	wm8974_write(codec, WM8974_IFACE, iface);
-+	wm8974_write(codec, WM8974_ADD, adn);
-+	return 0;
-+}
-+
-+static int wm8974_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
-+
-+	if(mute)
-+		wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
-+	else
-+		wm8974_write(codec, WM8974_DAC, mute_reg);
-+	return 0;
-+}
-+
-+/* liam need to make this lower power with dapm */
-+static int wm8974_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, clk and osc on, dac unmute, active */
-+		wm8974_write(codec, WM8974_POWER1, 0x1ff);
-+		wm8974_write(codec, WM8974_POWER2, 0x1ff);
-+		wm8974_write(codec, WM8974_POWER3, 0x1ff);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, dac mute, inactive */
-+
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		wm8974_write(codec, WM8974_POWER1, 0x0);
-+		wm8974_write(codec, WM8974_POWER2, 0x0);
-+		wm8974_write(codec, WM8974_POWER3, 0x0);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8974_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000)
-+
-+#define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8974_dai = {
-+	.name = "WM8974 HiFi",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8974_RATES,
-+		.formats = WM8974_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8974_RATES,
-+		.formats = WM8974_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8974_pcm_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8974_mute,
-+		.set_fmt = wm8974_set_dai_fmt,
-+		.set_clkdiv = wm8974_set_dai_clkdiv,
-+		.set_pll = wm8974_set_dai_pll,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8974_dai);
-+
-+static int wm8974_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8974_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8974_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8974 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8974_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "WM8974";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8974_read_reg_cache;
-+	codec->write = wm8974_write;
-+	codec->dapm_event = wm8974_dapm_event;
-+	codec->dai = &wm8974_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8974_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8974_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8974_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8974_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8974_reg);
-+
-+	wm8974_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "wm8974: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8974_add_controls(codec);
-+	wm8974_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8974: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8974_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8974 2 wire address is 0x1a
-+ */
-+#define I2C_DRIVERID_WM8974 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8974_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8974_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8974_socdev;
-+	struct wm8974_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if(ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8974_init(socdev);
-+	if(ret < 0) {
-+		err("failed to initialise WM8974\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int wm8974_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8974_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8974_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8974_i2c_driver = {
-+	.driver = {
-+		.name = "WM8974 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8974,
-+	.attach_adapter = wm8974_i2c_attach,
-+	.detach_client =  wm8974_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8974",
-+	.driver = &wm8974_i2c_driver,
-+};
-+#endif
-+
-+static int wm8974_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8974_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("WM8974 Audio Codec %s", WM8974_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8974_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8974_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8974_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8974_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8974 = {
-+	.probe = 	wm8974_probe,
-+	.remove = 	wm8974_remove,
-+	.suspend = 	wm8974_suspend,
-+	.resume =	wm8974_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974);
-+
-+MODULE_DESCRIPTION("ASoC WM8974 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8974.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8974.h
-@@ -0,0 +1,104 @@
-+/*
-+ * wm8974.h  --  WM8974 Soc Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8974_H
-+#define _WM8974_H
-+
-+/* WM8974 register space */
-+
-+#define WM8974_RESET		0x0
-+#define WM8974_POWER1		0x1
-+#define WM8974_POWER2		0x2
-+#define WM8974_POWER3		0x3
-+#define WM8974_IFACE		0x4
-+#define WM8974_COMP			0x5
-+#define WM8974_CLOCK		0x6
-+#define WM8974_ADD			0x7
-+#define WM8974_GPIO			0x8
-+#define WM8974_DAC			0xa
-+#define WM8974_DACVOL		0xb
-+#define WM8974_ADC			0xe
-+#define WM8974_ADCVOL		0xf
-+#define WM8974_EQ1			0x12
-+#define WM8974_EQ2			0x13
-+#define WM8974_EQ3			0x14
-+#define WM8974_EQ4			0x15
-+#define WM8974_EQ5			0x16
-+#define WM8974_DACLIM1		0x18
-+#define WM8974_DACLIM2		0x19
-+#define WM8974_NOTCH1		0x1b
-+#define WM8974_NOTCH2		0x1c
-+#define WM8974_NOTCH3		0x1d
-+#define WM8974_NOTCH4		0x1e
-+#define WM8974_ALC1			0x20
-+#define WM8974_ALC2			0x21
-+#define WM8974_ALC3			0x22
-+#define WM8974_NGATE		0x23
-+#define WM8974_PLLN			0x24
-+#define WM8974_PLLK1		0x25
-+#define WM8974_PLLK2		0x26
-+#define WM8974_PLLK3		0x27
-+#define WM8974_ATTEN		0x28
-+#define WM8974_INPUT		0x2c
-+#define WM8974_INPPGA		0x2d
-+#define WM8974_ADCBOOST		0x2f
-+#define WM8974_OUTPUT		0x31
-+#define WM8974_SPKMIX		0x32
-+#define WM8974_SPKVOL		0x36
-+#define WM8974_MONOMIX		0x38
-+
-+#define WM8974_CACHEREGNUM 	57
-+
-+/* Clock divider Id's */
-+#define WM8974_OPCLKDIV		0
-+#define WM8974_MCLKDIV		1
-+#define WM8974_ADCCLK		2
-+#define WM8974_DACCLK		3
-+#define WM8974_BCLKDIV		4
-+
-+/* DAC clock dividers */
-+#define WM8974_DACCLK_F2	(1 << 3)
-+#define WM8974_DACCLK_F4	(0 << 3)
-+
-+/* ADC clock dividers */
-+#define WM8974_ADCCLK_F2	(1 << 3)
-+#define WM8974_ADCCLK_F4	(0 << 3)
-+
-+/* PLL Out dividers */
-+#define WM8974_OPCLKDIV_1	(0 << 4)
-+#define WM8974_OPCLKDIV_2	(1 << 4)
-+#define WM8974_OPCLKDIV_3	(2 << 4)
-+#define WM8974_OPCLKDIV_4	(3 << 4)
-+
-+/* BCLK clock dividers */
-+#define WM8974_BCLKDIV_1	(0 << 2)
-+#define WM8974_BCLKDIV_2	(1 << 2)
-+#define WM8974_BCLKDIV_4	(2 << 2)
-+#define WM8974_BCLKDIV_8	(3 << 2)
-+#define WM8974_BCLKDIV_16	(4 << 2)
-+#define WM8974_BCLKDIV_32	(5 << 2)
-+
-+/* MCLK clock dividers */
-+#define WM8974_MCLKDIV_1	(0 << 5)
-+#define WM8974_MCLKDIV_1_5	(1 << 5)
-+#define WM8974_MCLKDIV_2	(2 << 5)
-+#define WM8974_MCLKDIV_3	(3 << 5)
-+#define WM8974_MCLKDIV_4	(4 << 5)
-+#define WM8974_MCLKDIV_6	(5 << 5)
-+#define WM8974_MCLKDIV_8	(6 << 5)
-+#define WM8974_MCLKDIV_12	(7 << 5)
-+
-+
-+struct wm8974_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8974_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8974;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm9713.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm9713.c
-@@ -0,0 +1,1220 @@
-+/*
-+ * wm9713.c  --  ALSA Soc WM9713 codec support
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    4th Feb 2006   Initial version.
-+ *
-+ *  Features:-
-+ *
-+ *   o Support for AC97 Codec, Voice DAC and Aux DAC
-+ *   o Support for DAPM
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/ac97_codec.h>
-+#include <sound/initval.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include "wm9713.h"
-+
-+#define WM9713_VERSION "0.12"
-+
-+struct wm9713_priv {
-+	u32 pll_in; /* PLL input frequency */
-+	u32 pll_out; /* PLL output frequency */
-+};
-+
-+static unsigned int ac97_read(struct snd_soc_codec *codec,
-+	unsigned int reg);
-+static int ac97_write(struct snd_soc_codec *codec,
-+	unsigned int reg, unsigned int val);
-+
-+/*
-+ * WM9713 register cache
-+ * Reg 0x3c bit 15 is used by touch driver.
-+ */
-+static const u16 wm9713_reg[] = {
-+	0x6174, 0x8080, 0x8080, 0x8080, // 6
-+	0xc880, 0xe808, 0xe808, 0x0808, // e
-+	0x00da, 0x8000, 0xd600, 0xaaa0, // 16
-+	0xaaa0, 0xaaa0, 0x0000, 0x0000, // 1e
-+	0x0f0f, 0x0040, 0x0000, 0x7f00, // 26
-+	0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
-+	0x0000, 0xbb80, 0x0000, 0x4523, // 36
-+	0x0000, 0x2000, 0x7eff, 0xffff, // 3e
-+	0x0000, 0x0000, 0x0080, 0x0000, // 46
-+	0x0000, 0x0000, 0xfffe, 0xffff, // 4e
-+	0x0000, 0x0000, 0x0000, 0xfffe, // 56
-+	0x4000, 0x0000, 0x0000, 0x0000, // 5e
-+	0xb032, 0x3e00, 0x0000, 0x0000, // 66
-+	0x0000, 0x0000, 0x0000, 0x0000, // 6e
-+	0x0000, 0x0000, 0x0000, 0x0006, // 76
-+	0x0001, 0x0000, 0x574d, 0x4c13, // 7e
-+	0x0000, 0x0000, 0x0000 // virtual hp & mic mixers
-+};
-+
-+/* virtual HP mixers regs */
-+#define HPL_MIXER	0x80
-+#define HPR_MIXER	0x82
-+#define MICB_MUX	0x82
-+
-+static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
-+static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
-+static const char *wm9713_rec_src[] =
-+	{"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
-+	"Mono Out", "Zh"};
-+static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
-+static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
-+static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv",
-+	"Mono Vmid", "Inv Vmid"};
-+static const char *wm9713_spk_pga[] =
-+	{"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
-+	"Speaker Vmid", "Inv Vmid"};
-+static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
-+	"Headphone Vmid"};
-+static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
-+static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
-+static const char *wm9713_dac_inv[] =
-+	{"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
-+	"Headphone Mono", "NC", "Vmid"};
-+static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
-+static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
-+static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
-+static const char *wm9713_micb_select[] = {"MPB", "MPA"};
-+
-+static const struct soc_enum wm9713_enum[] = {
-+SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
-+SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
-+SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux),  /* record mux mono 2 */
-+SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src),  /* record mux left 3 */
-+SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src),  /* record mux right 4*/
-+SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
-+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
-+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
-+SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
-+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
-+SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
-+SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
-+};
-+
-+static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
-+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
-+SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
-+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
-+SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE,15, 7, 1, 1),
-+SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
-+SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
-+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
-+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
-+
-+SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
-+SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
-+
-+SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
-+SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
-+SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
-+SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
-+
-+SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
-+SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
-+SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
-+
-+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
-+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
-+SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
-+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
-+SOC_ENUM("ALC Function", wm9713_enum[6]),
-+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
-+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
-+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
-+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
-+SOC_ENUM("ALC NG Type", wm9713_enum[17]),
-+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
-+
-+SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
-+SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
-+
-+SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
-+SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
-+SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
-+
-+SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
-+SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
-+SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
-+
-+SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
-+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
-+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
-+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
-+
-+SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
-+SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
-+SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
-+
-+SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
-+SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
-+SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
-+
-+SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
-+SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
-+SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
-+
-+SOC_ENUM("Bass Control", wm9713_enum[16]),
-+SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
-+SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
-+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
-+SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
-+SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
-+
-+SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
-+SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
-+SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
-+};
-+
-+/* add non dapm controls */
-+static int wm9713_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm9713_snd_ac97_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+	return 0;
-+}
-+
-+/* We have to create a fake left and right HP mixers because
-+ * the codec only has a single control that is shared by both channels.
-+ * This makes it impossible to determine the audio path using the current
-+ * register map, thus we add a new (virtual) register to help determine the
-+ * audio route within the device.
-+ */
-+static int mixer_event (struct snd_soc_dapm_widget *w, int event)
-+{
-+	u16 l, r, beep, tone, phone, rec, pcm, aux;
-+
-+	l = ac97_read(w->codec, HPL_MIXER);
-+	r = ac97_read(w->codec, HPR_MIXER);
-+	beep = ac97_read(w->codec, AC97_PC_BEEP);
-+	tone = ac97_read(w->codec, AC97_MASTER_TONE);
-+	phone = ac97_read(w->codec, AC97_PHONE);
-+	rec = ac97_read(w->codec, AC97_REC_SEL);
-+	pcm = ac97_read(w->codec, AC97_PCM);
-+	aux = ac97_read(w->codec, AC97_AUX);
-+
-+	if (event & SND_SOC_DAPM_PRE_REG)
-+		return 0;
-+	if (l & 0x1 || r & 0x1)
-+		ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
-+
-+	if (l & 0x2 || r & 0x2)
-+		ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
-+
-+	if (l & 0x4 || r & 0x4)
-+		ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
-+
-+	if (l & 0x8 || r & 0x8)
-+		ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
-+
-+	if (l & 0x10 || r & 0x10)
-+		ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
-+
-+	if (l & 0x20 || r & 0x20)
-+		ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
-+	else
-+		ac97_write(w->codec, AC97_AUX, aux | 0x8000);
-+
-+	return 0;
-+}
-+
-+/* Left Headphone Mixers */
-+static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
-+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
-+SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
-+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
-+};
-+
-+/* Right Headphone Mixers */
-+static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
-+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
-+SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
-+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
-+};
-+
-+/* headphone capture mux */
-+static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[1]);
-+
-+/* headphone mic mux */
-+static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[0]);
-+
-+/* Speaker Mixer */
-+static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
-+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
-+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
-+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
-+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
-+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
-+};
-+
-+/* Mono Mixer */
-+static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
-+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
-+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
-+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
-+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
-+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
-+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
-+SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
-+SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
-+};
-+
-+/* mono mic mux */
-+static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[2]);
-+
-+/* mono output mux */
-+static const struct snd_kcontrol_new wm9713_mono_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[7]);
-+
-+/* speaker left output mux */
-+static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[8]);
-+
-+/* speaker right output mux */
-+static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[9]);
-+
-+/* headphone left output mux */
-+static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[10]);
-+
-+/* headphone right output mux */
-+static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[11]);
-+
-+/* Out3 mux */
-+static const struct snd_kcontrol_new wm9713_out3_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[12]);
-+
-+/* Out4 mux */
-+static const struct snd_kcontrol_new wm9713_out4_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[13]);
-+
-+/* DAC inv mux 1 */
-+static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[14]);
-+
-+/* DAC inv mux 2 */
-+static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[15]);
-+
-+/* Capture source left */
-+static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[3]);
-+
-+/* Capture source right */
-+static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[4]);
-+
-+/* mic source */
-+static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[18]);
-+
-+/* mic source B virtual control */
-+static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
-+SOC_DAPM_ENUM("Route", wm9713_enum[19]);
-+
-+static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
-+SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hp_rec_mux_controls),
-+SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hp_mic_mux_controls),
-+SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_mono_mic_mux_controls),
-+SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_mono_mux_controls),
-+SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hp_spkl_mux_controls),
-+SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hp_spkr_mux_controls),
-+SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hpl_out_mux_controls),
-+SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_hpr_out_mux_controls),
-+SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_out3_mux_controls),
-+SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
-+	&wm9713_out4_mux_controls),
-+SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
-+	&wm9713_dac_inv1_mux_controls),
-+SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
-+	&wm9713_dac_inv2_mux_controls),
-+SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
-+	&wm9713_rec_srcl_mux_controls),
-+SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
-+	&wm9713_rec_srcr_mux_controls),
-+SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
-+	&wm9713_mic_sel_mux_controls ),
-+SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
-+	&wm9713_micb_sel_mux_controls ),
-+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
-+	&wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
-+	mixer_event, SND_SOC_DAPM_POST_REG),
-+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
-+	&wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
-+	mixer_event, SND_SOC_DAPM_POST_REG),
-+SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
-+	&wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
-+	&wm9713_speaker_mixer_controls[0],
-+	ARRAY_SIZE(wm9713_speaker_mixer_controls)),
-+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
-+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
-+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-+SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-+SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
-+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
-+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1),
-+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1),
-+SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
-+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
-+SND_SOC_DAPM_OUTPUT("MONO"),
-+SND_SOC_DAPM_OUTPUT("HPL"),
-+SND_SOC_DAPM_OUTPUT("HPR"),
-+SND_SOC_DAPM_OUTPUT("SPKL"),
-+SND_SOC_DAPM_OUTPUT("SPKR"),
-+SND_SOC_DAPM_OUTPUT("OUT3"),
-+SND_SOC_DAPM_OUTPUT("OUT4"),
-+SND_SOC_DAPM_INPUT("LINEL"),
-+SND_SOC_DAPM_INPUT("LINER"),
-+SND_SOC_DAPM_INPUT("MONOIN"),
-+SND_SOC_DAPM_INPUT("PCBEEP"),
-+SND_SOC_DAPM_INPUT("MIC1"),
-+SND_SOC_DAPM_INPUT("MIC2A"),
-+SND_SOC_DAPM_INPUT("MIC2B"),
-+SND_SOC_DAPM_VMID("VMID"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* left HP mixer */
-+	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
-+	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
-+	{"Left HP Mixer", "Aux Playback Switch",     "Aux DAC"},
-+	{"Left HP Mixer", "Bypass Playback Switch",  "Left Line In"},
-+	{"Left HP Mixer", "PCM Playback Switch",     "Left DAC"},
-+	{"Left HP Mixer", "MonoIn Playback Switch",  "Mono In"},
-+	{"Left HP Mixer", NULL,  "Capture Headphone Mux"},
-+
-+	/* right HP mixer */
-+	{"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
-+	{"Right HP Mixer", "Voice Playback Switch",   "Voice DAC"},
-+	{"Right HP Mixer", "Aux Playback Switch",     "Aux DAC"},
-+	{"Right HP Mixer", "Bypass Playback Switch",  "Right Line In"},
-+	{"Right HP Mixer", "PCM Playback Switch",     "Right DAC"},
-+	{"Right HP Mixer", "MonoIn Playback Switch",  "Mono In"},
-+	{"Right HP Mixer", NULL,  "Capture Headphone Mux"},
-+
-+	/* virtual mixer - mixes left & right channels for spk and mono */
-+	{"AC97 Mixer", NULL, "Left DAC"},
-+	{"AC97 Mixer", NULL, "Right DAC"},
-+	{"Line Mixer", NULL, "Right Line In"},
-+	{"Line Mixer", NULL, "Left Line In"},
-+	{"HP Mixer", NULL, "Left HP Mixer"},
-+	{"HP Mixer", NULL, "Right HP Mixer"},
-+	{"Capture Mixer", NULL, "Left Capture Source"},
-+	{"Capture Mixer", NULL, "Right Capture Source"},
-+
-+	/* speaker mixer */
-+	{"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
-+	{"Speaker Mixer", "Voice Playback Switch",   "Voice DAC"},
-+	{"Speaker Mixer", "Aux Playback Switch",     "Aux DAC"},
-+	{"Speaker Mixer", "Bypass Playback Switch",  "Line Mixer"},
-+	{"Speaker Mixer", "PCM Playback Switch",     "AC97 Mixer"},
-+	{"Speaker Mixer", "MonoIn Playback Switch",  "Mono In"},
-+
-+	/* mono mixer */
-+	{"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
-+	{"Mono Mixer", "Voice Playback Switch",   "Voice DAC"},
-+	{"Mono Mixer", "Aux Playback Switch",     "Aux DAC"},
-+	{"Mono Mixer", "Bypass Playback Switch",  "Line Mixer"},
-+	{"Mono Mixer", "PCM Playback Switch",     "AC97 Mixer"},
-+	{"Mono Mixer", NULL,  "Capture Mono Mux"},
-+
-+	/* DAC inv mux 1 */
-+	{"DAC Inv Mux 1", "Mono", "Mono Mixer"},
-+	{"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
-+	{"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
-+	{"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
-+	{"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
-+
-+	/* DAC inv mux 2 */
-+	{"DAC Inv Mux 2", "Mono", "Mono Mixer"},
-+	{"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
-+	{"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
-+	{"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
-+	{"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
-+
-+	/* headphone left mux */
-+	{"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
-+
-+	/* headphone right mux */
-+	{"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
-+
-+	/* speaker left mux */
-+	{"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
-+	{"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
-+	{"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
-+
-+	/* speaker right mux */
-+	{"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
-+	{"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
-+	{"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
-+
-+	/* mono mux */
-+	{"Mono Out Mux", "Mono", "Mono Mixer"},
-+	{"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
-+
-+	/* out 3 mux */
-+	{"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
-+
-+	/* out 4 mux */
-+	{"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
-+
-+	/* output pga */
-+	{"HPL", NULL, "Left Headphone"},
-+	{"Left Headphone", NULL, "Left Headphone Out Mux"},
-+	{"HPR", NULL, "Right Headphone"},
-+	{"Right Headphone", NULL, "Right Headphone Out Mux"},
-+	{"OUT3", NULL, "Out 3"},
-+	{"Out 3", NULL, "Out 3 Mux"},
-+	{"OUT4", NULL, "Out 4"},
-+	{"Out 4", NULL, "Out 4 Mux"},
-+	{"SPKL", NULL, "Left Speaker"},
-+	{"Left Speaker", NULL, "Left Speaker Out Mux"},
-+	{"SPKR", NULL, "Right Speaker"},
-+	{"Right Speaker", NULL, "Right Speaker Out Mux"},
-+	{"MONO", NULL, "Mono Out"},
-+	{"Mono Out", NULL, "Mono Out Mux"},
-+
-+	/* input pga */
-+	{"Left Line In", NULL, "LINEL"},
-+	{"Right Line In", NULL, "LINER"},
-+	{"Mono In", NULL, "MONOIN"},
-+	{"Mic A PGA", NULL, "Mic A Pre Amp"},
-+	{"Mic B PGA", NULL, "Mic B Pre Amp"},
-+
-+	/* left capture select */
-+	{"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
-+	{"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
-+	{"Left Capture Source", "Line", "LINEL"},
-+	{"Left Capture Source", "Mono In", "MONOIN"},
-+	{"Left Capture Source", "Headphone", "Left HP Mixer"},
-+	{"Left Capture Source", "Speaker", "Speaker Mixer"},
-+	{"Left Capture Source", "Mono Out", "Mono Mixer"},
-+
-+	/* right capture select */
-+	{"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
-+	{"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
-+	{"Right Capture Source", "Line", "LINER"},
-+	{"Right Capture Source", "Mono In", "MONOIN"},
-+	{"Right Capture Source", "Headphone", "Right HP Mixer"},
-+	{"Right Capture Source", "Speaker", "Speaker Mixer"},
-+	{"Right Capture Source", "Mono Out", "Mono Mixer"},
-+
-+	/* left ADC */
-+	{"Left ADC", NULL, "Left Capture Source"},
-+
-+	/* right ADC */
-+	{"Right ADC", NULL, "Right Capture Source"},
-+
-+	/* mic */
-+	{"Mic A Pre Amp", NULL, "Mic A Source"},
-+	{"Mic A Source", "Mic 1", "MIC1"},
-+	{"Mic A Source", "Mic 2 A", "MIC2A"},
-+	{"Mic A Source", "Mic 2 B", "Mic B Source"},
-+	{"Mic B Pre Amp", "MPB", "Mic B Source"},
-+	{"Mic B Source", NULL, "MIC2B"},
-+
-+	/* headphone capture */
-+	{"Capture Headphone Mux", "Stereo", "Capture Mixer"},
-+	{"Capture Headphone Mux", "Left", "Left Capture Source"},
-+	{"Capture Headphone Mux", "Right", "Right Capture Source"},
-+
-+	/* mono capture */
-+	{"Capture Mono Mux", "Stereo", "Capture Mixer"},
-+	{"Capture Mono Mux", "Left", "Left Capture Source"},
-+	{"Capture Mono Mux", "Right", "Right Capture Source"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm9713_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+static unsigned int ac97_read(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+
-+	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
-+		reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
-+		reg == AC97_CD)
-+		return soc_ac97_ops.read(codec->ac97, reg);
-+	else {
-+		reg = reg >> 1;
-+
-+		if (reg > (ARRAY_SIZE(wm9713_reg)))
-+			return -EIO;
-+
-+		return cache[reg];
-+	}
-+}
-+
-+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int val)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg < 0x7c)
-+		soc_ac97_ops.write(codec->ac97, reg, val);
-+	reg = reg >> 1;
-+	if (reg <= (ARRAY_SIZE(wm9713_reg)))
-+		cache[reg] = val;
-+
-+	return 0;
-+}
-+
-+struct pll_ {
-+	unsigned int in_hz;
-+	unsigned int lf:1; /* allows low frequency use */
-+	unsigned int sdm:1; /* allows fraction n div */
-+	unsigned int divsel:1; /* enables input clock div */
-+	unsigned int divctl:1; /* input clock divider */
-+	unsigned int n:4;
-+	unsigned int k;
-+};
-+
-+struct pll_ pll[] = {
-+	{13000000, 0, 1, 0, 0, 7, 0x23f488},
-+	{2048000,  1, 0, 0, 0, 12, 0x0},
-+	{4096000,  1, 0, 0, 0, 6, 0x0},
-+	{12288000, 0, 0, 0, 0, 8, 0x0},
-+	/* liam - add more entries */
-+};
-+
-+static int wm9713_set_pll(struct snd_soc_codec *codec,
-+	int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct wm9713_priv *wm9713 = codec->private_data;
-+	int i;
-+	u16 reg, reg2;
-+
-+	/* turn PLL off ? */
-+	if (freq_in == 0 || freq_out == 0) {
-+		/* disable PLL power and select ext source */
-+		reg = ac97_read(codec, AC97_HANDSET_RATE);
-+		ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
-+		reg = ac97_read(codec, AC97_EXTENDED_MID);
-+		ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
-+		wm9713->pll_out = 0;
-+		return 0;
-+	}
-+
-+	for (i = 0; i < ARRAY_SIZE(pll); i++) {
-+		if (pll[i].in_hz == freq_in)
-+			goto found;
-+	}
-+	return -EINVAL;
-+
-+found:
-+	if (pll[i].sdm == 0) {
-+		reg = (pll[i].n << 12) | (pll[i].lf << 11) |
-+			(pll[i].divsel << 9) | (pll[i].divctl << 8);
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+	} else {
-+		/* write the fractional k to the reg 0x46 pages */
-+		reg2 = (pll[i].n << 12) | (pll[i].lf << 11) | (pll[i].sdm << 10) |
-+			(pll[i].divsel << 9) | (pll[i].divctl << 8);
-+
-+		reg = reg2 | (0x5 << 4) | (pll[i].k >> 20); /* K [21:20] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+
-+		reg = reg2 | (0x4 << 4) | ((pll[i].k >> 16) & 0xf); /* K [19:16] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+
-+		reg = reg2 | (0x3 << 4) | ((pll[i].k >> 12) & 0xf); /* K [15:12] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+
-+		reg = reg2 | (0x2 << 4) | ((pll[i].k >> 8) & 0xf); /* K [11:8] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+
-+		reg = reg2 | (0x1 << 4) | ((pll[i].k >> 4) & 0xf); /* K [7:4] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+
-+		reg = reg2 | (0x0 << 4) | (pll[i].k & 0xf); /* K [3:0] */
-+		ac97_write(codec, AC97_LINE1_LEVEL, reg);
-+	}
-+
-+	/* turn PLL on and select as source */
-+	reg = ac97_read(codec, AC97_EXTENDED_MID);
-+	ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
-+	reg = ac97_read(codec, AC97_HANDSET_RATE);
-+	ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
-+	wm9713->pll_out = freq_out;
-+	wm9713->pll_in = freq_in;
-+
-+	/* wait 10ms AC97 link frames for the link to stabilise */
-+	schedule_timeout_interruptible(msecs_to_jiffies(10));
-+	return 0;
-+}
-+
-+static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
-+}
-+
-+/*
-+ * Tristate the PCM DAI lines, tristate can be disabled by calling
-+ * wm9713_set_dai_fmt()
-+ */
-+static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
-+	int tristate)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff;
-+
-+	if (tristate)
-+		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
-+
-+	return 0;
-+}
-+
-+/*
-+ * Configure WM9713 clock dividers.
-+ * Voice DAC needs 256 FS
-+ */
-+static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM9713_PCMCLK_DIV:
-+		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
-+		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
-+		break;
-+	case WM9713_CLKA_MULT:
-+		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd;
-+		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
-+		break;
-+	case WM9713_CLKB_MULT:
-+		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb;
-+		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
-+		break;
-+	case WM9713_HIFI_DIV:
-+		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff;
-+		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
-+		break;
-+	case WM9713_PCMBCLK_DIV:
-+		reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff;
-+		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+};
-+
-+static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffe2;
-+	u16 reg = 0x8000;
-+
-+	/* clock masters */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK){
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		reg |= 0x4000;
-+		gpio |= 0x0008;
-+		break;
-+	case SND_SOC_DAIFMT_CBM_CFS:
-+		reg |= 0x6000;
-+		gpio |= 0x000c;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		reg |= 0x0200;
-+		gpio |= 0x000d;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFM:
-+		gpio |= 0x0009;
-+		break;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_IB_IF:
-+		reg |= 0x00c0;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		reg |= 0x0080;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		reg |= 0x0040;
-+		break;
-+	}
-+
-+	/* DAI format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		reg |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		reg |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		reg |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		reg |= 0x0043;
-+		break;
-+	}
-+
-+	ac97_write(codec, AC97_GPIO_CFG, gpio);
-+	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
-+	return 0;
-+}
-+
-+static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
-+
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		reg |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		reg |= 0x0008;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		reg |= 0x000c;
-+		break;
-+	}
-+
-+	/* enable PCM interface in master mode */
-+	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
-+	return 0;
-+}
-+
-+static void wm9713_voiceshutdown(snd_pcm_substream_t *substream)
-+{
-+    struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+    struct snd_soc_device *socdev = rtd->socdev;
-+    struct snd_soc_codec *codec = socdev->codec;
-+    u16 status;
-+
-+    /* Gracefully shut down the voice interface. */
-+    status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-+    ac97_write(codec,AC97_HANDSET_RATE,0x0280);
-+    schedule_timeout_interruptible(msecs_to_jiffies(1));
-+    ac97_write(codec,AC97_HANDSET_RATE,0x0F80);
-+    ac97_write(codec,AC97_EXTENDED_MID,status);
-+}
-+
-+static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg;
-+	u16 vra;
-+
-+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
-+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+		reg = AC97_PCM_FRONT_DAC_RATE;
-+	else
-+		reg = AC97_PCM_LR_ADC_RATE;
-+
-+	return ac97_write(codec, reg, runtime->rate);
-+}
-+
-+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 vra, xsle;
-+
-+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
-+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
-+	xsle = ac97_read(codec, AC97_PCI_SID);
-+	ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
-+
-+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
-+		return -ENODEV;
-+
-+	return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
-+}
-+
-+#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-+
-+#define WM9713_PCM_FORMATS \
-+	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
-+	 SNDRV_PCM_FORMAT_S24_LE)
-+
-+struct snd_soc_codec_dai wm9713_dai[] = {
-+{
-+	.name = "AC97 HiFi",
-+	.playback = {
-+		.stream_name = "HiFi Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM9713_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.stream_name = "HiFi Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM9713_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.prepare = ac97_hifi_prepare,},
-+	},
-+	{
-+	.name = "AC97 Aux",
-+	.playback = {
-+		.stream_name = "Aux Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM9713_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.prepare = ac97_aux_prepare,},
-+	},
-+	{
-+	.name = "WM9713 Voice",
-+	.playback = {
-+		.stream_name = "Voice Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM9713_RATES,
-+		.formats = WM9713_PCM_FORMATS,},
-+	.capture = {
-+		.stream_name = "Voice Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM9713_RATES,
-+		.formats = WM9713_PCM_FORMATS,},
-+	.ops = {
-+		.hw_params = wm9713_pcm_hw_params,
-+		.shutdown = wm9713_voiceshutdown,},
-+	.dai_ops = {
-+		.set_clkdiv = wm9713_set_dai_clkdiv,
-+		.set_pll = wm9713_set_dai_pll,
-+		.set_fmt = wm9713_set_dai_fmt,
-+		.set_tristate = wm9713_set_dai_tristate,
-+	},
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm9713_dai);
-+
-+int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
-+{
-+	if (try_warm && soc_ac97_ops.warm_reset) {
-+		soc_ac97_ops.warm_reset(codec->ac97);
-+		if (!(ac97_read(codec, 0) & 0x8000))
-+			return 1;
-+	}
-+
-+	soc_ac97_ops.reset(codec->ac97);
-+	if (ac97_read(codec, 0) & 0x8000)
-+		return -EIO;
-+	return 0;
-+}
-+EXPORT_SYMBOL_GPL(wm9713_reset);
-+
-+static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 reg;
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* enable thermal shutdown */
-+		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
-+		ac97_write(codec, AC97_EXTENDED_MID, reg);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* enable master bias and vmid */
-+		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
-+		ac97_write(codec, AC97_EXTENDED_MID, reg);
-+		ac97_write(codec, AC97_POWERDOWN, 0x0000);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* disable everything including AC link */
-+		ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
-+		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
-+		ac97_write(codec, AC97_POWERDOWN, 0xffff);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+static int wm9713_soc_suspend(struct platform_device *pdev,
-+	pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm9713_soc_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm9713_priv *wm9713 = codec->private_data;
-+	int i, ret;
-+	u16 *cache = codec->reg_cache;
-+
-+	if ((ret = wm9713_reset(codec, 1)) < 0){
-+		printk(KERN_ERR "could not reset AC97 codec\n");
-+		return ret;
-+	}
-+
-+	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* do we need to re-start the PLL ? */
-+	if (wm9713->pll_out)
-+		wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
-+
-+	/* only synchronise the codec if warm reset failed */
-+	if (ret == 0) {
-+		for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i+=2) {
-+			if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
-+				i == AC97_EXTENDED_MSTATUS || i > 0x66)
-+				continue;
-+			soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
-+		}
-+	}
-+
-+	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
-+		wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
-+
-+	return ret;
-+}
-+
-+static int wm9713_soc_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec;
-+	int ret = 0, reg;
-+
-+	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
-+
-+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (socdev->codec == NULL)
-+		return -ENOMEM;
-+	codec = socdev->codec;
-+	mutex_init(&codec->mutex);
-+
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm9713_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL){
-+		ret = -ENOMEM;
-+		goto cache_err;
-+	}
-+	memcpy(codec->reg_cache, wm9713_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm9713_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9713_reg);
-+	codec->reg_cache_step = 2;
-+
-+	codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
-+	if (codec->private_data == NULL) {
-+		ret = -ENOMEM;
-+		goto priv_err;
-+	}
-+
-+	codec->name = "WM9713";
-+	codec->owner = THIS_MODULE;
-+	codec->dai = wm9713_dai;
-+	codec->num_dai = ARRAY_SIZE(wm9713_dai);
-+	codec->write = ac97_write;
-+	codec->read = ac97_read;
-+	codec->dapm_event = wm9713_dapm_event;
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
-+	if (ret < 0)
-+		goto codec_err;
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0)
-+		goto pcm_err;
-+
-+	/* do a cold reset for the controller and then try
-+	 * a warm reset followed by an optional cold reset for codec */
-+	wm9713_reset(codec, 0);
-+	ret = wm9713_reset(codec, 1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "AC97 link error\n");
-+		goto reset_err;
-+	}
-+
-+	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* unmute the adc - move to kcontrol */
-+	reg = ac97_read(codec, AC97_CD) & 0x7fff;
-+	ac97_write(codec, AC97_CD, reg);
-+
-+	wm9713_add_controls(codec);
-+	wm9713_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0)
-+		goto reset_err;
-+	return 0;
-+
-+reset_err:
-+	snd_soc_free_pcms(socdev);
-+
-+pcm_err:
-+	snd_soc_free_ac97_codec(codec);
-+
-+codec_err:
-+	kfree(codec->private_data);
-+
-+priv_err:
-+	kfree(codec->reg_cache);
-+
-+cache_err:
-+	kfree(socdev->codec);
-+	socdev->codec = NULL;
-+	return ret;
-+}
-+
-+static int wm9713_soc_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec == NULL)
-+		return 0;
-+
-+	snd_soc_dapm_free(socdev);
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_free_ac97_codec(codec);
-+	kfree(codec->private_data);
-+	kfree(codec->reg_cache);
-+	kfree(codec->dai);
-+	kfree(codec);
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm9713 = {
-+	.probe = 	wm9713_soc_probe,
-+	.remove = 	wm9713_soc_remove,
-+	.suspend =	wm9713_soc_suspend,
-+	.resume = 	wm9713_soc_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713);
-+
-+MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm9713.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm9713.h
-@@ -0,0 +1,51 @@
-+/*
-+ * wm9713.h  --  WM9713 Soc Audio driver
-+ */
-+
-+#ifndef _WM9713_H
-+#define _WM9713_H
-+
-+/* clock inputs */
-+#define WM9713_CLKA_PIN			0
-+#define WM9713_CLKB_PIN			1
-+
-+/* clock divider ID's */
-+#define WM9713_PCMCLK_DIV		0
-+#define WM9713_CLKA_MULT		1
-+#define WM9713_CLKB_MULT		2
-+#define WM9713_HIFI_DIV			3
-+#define WM9713_PCMBCLK_DIV		4
-+
-+/* PCM clk div */
-+#define WM9713_PCMDIV(x)	((x - 1) << 8)
-+
-+/* HiFi Div */
-+#define WM9713_HIFIDIV(x)	((x - 1) << 12)
-+
-+/* MCLK clock mulitipliers */
-+#define WM9713_CLKA_X1		(0 << 1)
-+#define WM9713_CLKA_X2		(1 << 1)
-+#define WM9713_CLKB_X1		(0 << 2)
-+#define WM9713_CLKB_X2		(1 << 2)
-+
-+/* MCLK clock MUX */
-+#define WM9713_CLK_MUX_A		(0 << 0)
-+#define WM9713_CLK_MUX_B		(1 << 0)
-+
-+/* Voice DAI BCLK divider */
-+#define WM9713_PCMBCLK_DIV_1	(0 << 9)
-+#define WM9713_PCMBCLK_DIV_2	(1 << 9)
-+#define WM9713_PCMBCLK_DIV_4	(2 << 9)
-+#define WM9713_PCMBCLK_DIV_8	(3 << 9)
-+#define WM9713_PCMBCLK_DIV_16	(4 << 9)
-+
-+#define WM9713_DAI_AC97_HIFI	0
-+#define WM9713_DAI_AC97_AUX		1
-+#define WM9713_DAI_PCM_VOICE	2
-+
-+extern struct snd_soc_codec_device soc_codec_dev_wm9713;
-+extern struct snd_soc_codec_dai wm9713_dai[3];
-+
-+int wm9713_reset(struct snd_soc_codec *codec,  int try_warm);
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone.c
-@@ -0,0 +1,127 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    30th Oct 2005   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/ac97.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-ac97.h"
-+
-+static struct snd_soc_machine mainstone;
-+static long mst_audio_suspend_mask;
-+
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	mst_audio_suspend_mask = MST_MSCWR2;
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static struct snd_soc_machine_config codecs[] = {
-+{
-+	.name = "AC97",
-+	.sname = "AC97 HiFi",
-+	.iface = &pxa_ac97_interface[0],
-+},
-+{
-+	.name = "AC97 Aux",
-+	.sname = "AC97 Aux",
-+	.iface = &pxa_ac97_interface[1],
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.config = codecs,
-+	.nconfigs = ARRAY_SIZE(codecs),
-+};
-+
-+static struct snd_soc_device mainstone_snd_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_ac97,
-+};
-+
-+static struct platform_device *mainstone_snd_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
-+	mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
-+	ret = platform_device_add(mainstone_snd_device);
-+
-+	if (ret)
-+		platform_device_put(mainstone_snd_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_baseband.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_baseband.c
-@@ -0,0 +1,212 @@
-+/*
-+ * mainstone_baseband.c
-+ * Mainstone Example Baseband modem  --  ALSA Soc Audio Layer
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    15th Apr 2006   Initial version.
-+ *
-+ * This is example code to demonstrate connecting a baseband modem to the PCM
-+ * DAI on the WM9713 codec on the Intel Mainstone platform. It is by no means
-+ * complete as it requires code to control the modem.
-+ *
-+ * The architecture consists of the WM9713 AC97 DAI connected to the PXA27x
-+ * AC97 controller and the WM9713 PCM DAI connected to the basebands DAI. The
-+ * baseband is controlled via a serial port. Audio is routed between the PXA27x
-+ * and the baseband via internal WM9713 analog paths.
-+ *
-+ * This driver is not the baseband modem driver. This driver only calls
-+ * functions from the Baseband driver to set up it's PCM DAI.
-+ *
-+ * It's intended to use this driver as follows:-
-+ *
-+ *  1. open() WM9713 PCM audio device.
-+ *  2. open() serial device (for AT commands).
-+ *  3. configure PCM audio device (rate etc) - sets up WM9713 PCM DAI,
-+ *      this will also set up the baseband PCM DAI (via calling baseband driver).
-+ *  4. send any further AT commands to set up baseband.
-+ *  5. configure codec audio mixer paths.
-+ *  6. open(), configure and read/write AC97 audio device - to Tx/Rx voice
-+ *
-+ * The PCM audio device is opened but IO is never performed on it as the IO is
-+ * directly between the codec and the baseband (and not the CPU).
-+ *
-+ * TODO:
-+ *  o Implement callbacks
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/platform_device.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/hardware.h>
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/audio.h>
-+#include <asm/arch/ssp.h>
-+
-+#include "../codecs/wm9713.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-ac97.h"
-+#include "pxa2xx-ssp.h"
-+
-+static struct snd_soc_machine mainstone;
-+
-+/* Do specific baseband PCM voice startup here */
-+static int baseband_startup(struct snd_pcm_substream *substream)
-+{
-+	return 0;
-+}
-+
-+/* Do specific baseband PCM voice shutdown here */
-+static void baseband_shutdown (struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/* Do specific baseband modem PCM voice hw params init here */
-+static int baseband_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	return 0;
-+}
-+
-+/* Do specific baseband modem PCM voice hw params free here */
-+static int baseband_hw_free(struct snd_pcm_substream *substream)
-+{
-+	return 0;
-+}
-+
-+/*
-+ * Baseband Processor DAI
-+ */
-+static struct snd_soc_cpu_dai baseband_dai =
-+{	.name = "Baseband",
-+	.id = 0,
-+	.type = SND_SOC_DAI_PCM,
-+	.playback = {
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = SNDRV_PCM_RATE_8000,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = SNDRV_PCM_RATE_8000,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.startup = baseband_startup,
-+		.shutdown = baseband_shutdown,
-+		.hw_params = baseband_hw_params,
-+		.hw_free = baseband_hw_free,
-+		},
-+};
-+
-+/* PM */
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_wm9713_init(struct snd_soc_codec *codec)
-+{
-+	return 0;
-+}
-+
-+/* the physical audio connections between the WM9713, Baseband and pxa2xx */
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{
-+	.name = "AC97",
-+	.stream_name = "AC97 HiFi",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
-+	.init = mainstone_wm9713_init,
-+},
-+{
-+	.name = "AC97 Aux",
-+	.stream_name = "AC97 Aux",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
-+},
-+{
-+	.name = "Baseband",
-+	.stream_name = "Voice",
-+	.cpu_dai = &baseband_dai,
-+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct snd_soc_device mainstone_snd_ac97_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm9713,
-+};
-+
-+static struct platform_device *mainstone_snd_ac97_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_ac97_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
-+	mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
-+
-+	if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
-+		platform_device_put(mainstone_snd_ac97_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_ac97_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("Mainstone Example Baseband PCM Interface");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_bluetooth.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_bluetooth.c
-@@ -0,0 +1,371 @@
-+/*
-+ * mainstone_bluetooth.c
-+ * Mainstone Example Bluetooth  --  ALSA Soc Audio Layer
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    15th May 2006   Initial version.
-+ *
-+ * This is example code to demonstrate connecting a bluetooth codec to the PCM
-+ * DAI on the WM8753 codec on the Intel Mainstone platform. It is by no means
-+ * complete as it requires code to control the BT codec.
-+ *
-+ * The architecture consists of the WM8753 HIFI DAI connected to the PXA27x
-+ * I2S controller and the WM8753 PCM DAI connected to the bluetooth DAI. The
-+ * bluetooth codec and wm8753 are controlled via I2C. Audio is routed between
-+ * the PXA27x and the bluetooth via internal WM8753 analog paths.
-+ *
-+ * This example supports the following audio input/outputs.
-+ *
-+ *  o Board mounted Mic and Speaker (spk has amplifier)
-+ *  o Headphones via jack socket
-+ *  o BT source and sink
-+ *
-+ * This driver is not the bluetooth codec driver. This driver only calls
-+ * functions from the Bluetooth driver to set up it's PCM DAI.
-+ *
-+ * It's intended to use the driver as follows:-
-+ *
-+ *  1. open() WM8753 PCM audio device.
-+ *  2. configure PCM audio device (rate etc) - sets up WM8753 PCM DAI,
-+ *      this should also set up the BT codec DAI (via calling bt driver).
-+ *  3. configure codec audio mixer paths.
-+ *  4. open(), configure and read/write HIFI audio device - to Tx/Rx voice
-+ *
-+ * The PCM audio device is opened but IO is never performed on it as the IO is
-+ * directly between the codec and the BT codec (and not the CPU).
-+ *
-+ * TODO:
-+ *  o Implement callbacks
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/platform_device.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/hardware.h>
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/audio.h>
-+#include <asm/arch/ssp.h>
-+
-+#include "../codecs/wm8753.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-i2s.h"
-+#include "pxa2xx-ssp.h"
-+
-+static struct snd_soc_machine mainstone;
-+
-+/* Do specific bluetooth PCM startup here */
-+static int bt_startup(struct snd_pcm_substream *substream)
-+{
-+	return 0;
-+}
-+
-+/* Do specific bluetooth PCM shutdown here */
-+static void bt_shutdown (struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/* Do pecific bluetooth PCM hw params init here */
-+static int bt_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	return 0;
-+}
-+
-+/* Do specific bluetooth PCM hw params free here */
-+static int bt_hw_free(struct snd_pcm_substream *substream)
-+{
-+	return 0;
-+}
-+
-+#define BT_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100)
-+
-+/*
-+ * BT Codec DAI
-+ */
-+static struct snd_soc_cpu_dai bt_dai =
-+{	.name = "Bluetooth",
-+	.id = 0,
-+	.type = SND_SOC_DAI_PCM,
-+	.playback = {
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = BT_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = BT_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.startup = bt_startup,
-+		.shutdown = bt_shutdown,
-+		.hw_params = bt_hw_params,
-+		.hw_free = bt_hw_free,
-+		},
-+};
-+
-+/* PM */
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	return 0;
-+}
-+
-+/*
-+ * Machine audio functions.
-+ *
-+ * The machine now has 3 extra audio controls.
-+ *
-+ * Jack function: Sets function (device plugged into Jack) to nothing (Off)
-+ *                or Headphones.
-+ *
-+ * Mic function: Set the on board Mic to On or Off
-+ * Spk function: Set the on board Spk to On or Off
-+ *
-+ * example: BT playback (of far end) and capture (of near end)
-+ *  Set Mic and Speaker to On, open BT alsa interface as above and set up
-+ *  internal audio paths.
-+ */
-+
-+static int machine_jack_func = 0;
-+static int machine_spk_func = 0;
-+static int machine_mic_func = 0;
-+
-+static int machine_get_jack(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = machine_jack_func;
-+	return 0;
-+}
-+
-+static int machine_set_jack(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-+	machine_jack_func = ucontrol->value.integer.value[0];
-+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", machine_jack_func);
-+	return 0;
-+}
-+
-+static int machine_get_spk(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = machine_spk_func;
-+	return 0;
-+}
-+
-+static int machine_set_spk(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-+
-+	if (machine_spk_func == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	machine_spk_func = ucontrol->value.integer.value[0];
-+	snd_soc_dapm_set_endpoint(codec, "Spk", machine_spk_func);
-+	return 1;
-+}
-+
-+static int machine_get_mic(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = machine_spk_func;
-+	return 0;
-+}
-+
-+static int machine_set_mic(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-+
-+	if (machine_spk_func == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	machine_spk_func = ucontrol->value.integer.value[0];
-+	snd_soc_dapm_set_endpoint(codec, "Mic", machine_mic_func);
-+	return 1;
-+}
-+
-+/* turns on board speaker amp on/off */
-+static int machine_amp_event(struct snd_soc_dapm_widget *w, int event)
-+{
-+#if 0
-+	if (SND_SOC_DAPM_EVENT_ON(event))
-+		/* on */
-+	else
-+		/* off */
-+#endif
-+	return 0;
-+}
-+
-+/* machine dapm widgets */
-+static const struct snd_soc_dapm_widget machine_dapm_widgets[] = {
-+SND_SOC_DAPM_HP("Headphone Jack", NULL),
-+SND_SOC_DAPM_SPK("Spk", machine_amp_event),
-+SND_SOC_DAPM_MIC("Mic", NULL),
-+};
-+
-+/* machine connections to the codec pins */
-+static const char* audio_map[][3] = {
-+
-+	/* headphone connected to LOUT1, ROUT1 */
-+	{"Headphone Jack", NULL, "LOUT"},
-+	{"Headphone Jack", NULL, "ROUT"},
-+
-+	/* speaker connected to LOUT2, ROUT2 */
-+	{"Spk", NULL, "ROUT2"},
-+	{"Spk", NULL, "LOUT2"},
-+
-+	/* mic is connected to MIC1 (via Mic Bias) */
-+	{"MIC1", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+static const char* jack_function[] = {"Off", "Headphone"};
-+static const char* spk_function[] = {"Off", "On"};
-+static const char* mic_function[] = {"Off", "On"};
-+static const struct soc_enum machine_ctl_enum[] = {
-+	SOC_ENUM_SINGLE_EXT(2, jack_function),
-+	SOC_ENUM_SINGLE_EXT(2, spk_function),
-+	SOC_ENUM_SINGLE_EXT(2, mic_function),
-+};
-+
-+static const struct snd_kcontrol_new wm8753_machine_controls[] = {
-+	SOC_ENUM_EXT("Jack Function", machine_ctl_enum[0], machine_get_jack, machine_set_jack),
-+	SOC_ENUM_EXT("Speaker Function", machine_ctl_enum[1], machine_get_spk, machine_set_spk),
-+	SOC_ENUM_EXT("Mic Function", machine_ctl_enum[2], machine_get_mic, machine_set_mic),
-+};
-+
-+static int mainstone_wm8753_init(struct snd_soc_codec *codec)
-+{
-+	int i, err;
-+
-+	/* not used on this machine - e.g. will never be powered up */
-+	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
-+	snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
-+	snd_soc_dapm_set_endpoint(codec, "MONO2", 0);
-+	snd_soc_dapm_set_endpoint(codec, "MONO1", 0);
-+	snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
-+	snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
-+	snd_soc_dapm_set_endpoint(codec, "RXP", 0);
-+	snd_soc_dapm_set_endpoint(codec, "RXN", 0);
-+	snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
-+
-+	/* Add machine specific controls */
-+	for (i = 0; i < ARRAY_SIZE(wm8753_machine_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8753_machine_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	/* Add machine specific widgets */
-+	for(i = 0; i < ARRAY_SIZE(machine_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &machine_dapm_widgets[i]);
-+	}
-+
-+	/* Set up machine specific audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{ /* Hifi Playback - for similatious use with voice below */
-+	.name = "WM8753",
-+	.stream_name = "WM8753 HiFi",
-+	.cpu_dai = &pxa_i2s_dai,
-+	.codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
-+	.init = mainstone_wm8753_init,
-+},
-+{ /* Voice via BT */
-+	.name = "Bluetooth",
-+	.stream_name = "Voice",
-+	.cpu_dai = &bt_dai,
-+	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct snd_soc_device mainstone_snd_wm8753_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm8753,
-+};
-+
-+static struct platform_device *mainstone_snd_wm8753_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_wm8753_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_wm8753_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_wm8753_device, &mainstone_snd_wm8753_devdata);
-+	mainstone_snd_wm8753_devdata.dev = &mainstone_snd_wm8753_device->dev;
-+
-+	if((ret = platform_device_add(mainstone_snd_wm8753_device)) != 0)
-+		platform_device_put(mainstone_snd_wm8753_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_wm8753_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("Mainstone Example Bluetooth PCM Interface");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8731.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8731.c
-@@ -0,0 +1,203 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    5th June 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/wm8731.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-i2s.h"
-+
-+static struct snd_soc_machine mainstone;
-+
-+static int mainstone_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	unsigned int clk = 0;
-+	int ret = 0;
-+
-+	switch (params_rate(params)) {
-+	case 8000:
-+	case 16000:
-+	case 48000:
-+	case 96000:
-+		clk = 12288000;
-+		break;
-+	case 11025:
-+	case 22050:
-+	case 44100:
-+		clk = 11289600;
-+		break;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the codec system clock for DAC and ADC */
-+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the I2S system clock as input (unused) */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+static struct snd_soc_ops mainstone_ops = {
-+	.hw_params = mainstone_hw_params,
-+};
-+
-+static const struct snd_soc_dapm_widget dapm_widgets[] = {
-+	SND_SOC_DAPM_MIC("Int Mic", NULL),
-+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-+};
-+
-+static const char* intercon[][3] = {
-+
-+	/* speaker connected to LHPOUT */
-+	{"Ext Spk", NULL, "LHPOUT"},
-+
-+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
-+	{"MICIN", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Int Mic"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+/*
-+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
-+ */
-+static int mainstone_wm8731_init(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+
-+	/* Add specific widgets */
-+	for(i = 0; i < ARRAY_SIZE(dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &dapm_widgets[i]);
-+	}
-+
-+	/* Set up specific audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]);
-+	}
-+
-+	/* not connected */
-+	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
-+	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
-+
-+	/* always connected */
-+	snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
-+	snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{
-+	.name = "WM8731",
-+	.stream_name = "WM8731 HiFi",
-+	.cpu_dai = &pxa_i2s_dai,
-+	.codec_dai = &wm8731_dai,
-+	.init = mainstone_wm8731_init,
-+	.ops = &mainstone_ops,
-+	},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct wm8731_setup_data corgi_wm8731_setup = {
-+	.i2c_address = 0x1b,
-+};
-+
-+static struct snd_soc_device mainstone_snd_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm8731,
-+	.codec_data = &corgi_wm8731_setup,
-+};
-+
-+static struct platform_device *mainstone_snd_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
-+	mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
-+	ret = platform_device_add(mainstone_snd_device);
-+
-+	if (ret)
-+		platform_device_put(mainstone_snd_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM8731 Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8753.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8753.c
-@@ -0,0 +1,547 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    30th Oct 2005   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/hardware.h>
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/wm8753.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-i2s.h"
-+#include "pxa2xx-ssp.h"
-+
-+/*
-+ * SSP GPIO's
-+ */
-+#define GPIO26_SSP1RX_MD	(26 | GPIO_ALT_FN_1_IN)
-+#define GPIO25_SSP1TX_MD	(25 | GPIO_ALT_FN_2_OUT)
-+#define GPIO23_SSP1CLKS_MD	(23 | GPIO_ALT_FN_2_IN)
-+#define GPIO24_SSP1FRMS_MD	(24 | GPIO_ALT_FN_2_IN)
-+#define GPIO23_SSP1CLKM_MD	(23 | GPIO_ALT_FN_2_OUT)
-+#define GPIO24_SSP1FRMM_MD	(24 | GPIO_ALT_FN_2_OUT)
-+#define GPIO53_SSP1SYSCLK_MD	(53 | GPIO_ALT_FN_2_OUT)
-+
-+#define GPIO11_SSP2RX_MD	(11 | GPIO_ALT_FN_2_IN)
-+#define GPIO13_SSP2TX_MD	(13 | GPIO_ALT_FN_1_OUT)
-+#define GPIO22_SSP2CLKS_MD	(22 | GPIO_ALT_FN_3_IN)
-+#define GPIO88_SSP2FRMS_MD	(88 | GPIO_ALT_FN_3_IN)
-+#define GPIO22_SSP2CLKM_MD	(22 | GPIO_ALT_FN_3_OUT)
-+#define GPIO88_SSP2FRMM_MD	(88 | GPIO_ALT_FN_3_OUT)
-+#define GPIO22_SSP2SYSCLK_MD	(22 | GPIO_ALT_FN_2_OUT)
-+
-+#define GPIO82_SSP3RX_MD	(82 | GPIO_ALT_FN_1_IN)
-+#define GPIO81_SSP3TX_MD	(81 | GPIO_ALT_FN_1_OUT)
-+#define GPIO84_SSP3CLKS_MD	(84 | GPIO_ALT_FN_1_IN)
-+#define GPIO83_SSP3FRMS_MD	(83 | GPIO_ALT_FN_1_IN)
-+#define GPIO84_SSP3CLKM_MD	(84 | GPIO_ALT_FN_1_OUT)
-+#define GPIO83_SSP3FRMM_MD	(83 | GPIO_ALT_FN_1_OUT)
-+#define GPIO45_SSP3SYSCLK_MD	(45 | GPIO_ALT_FN_3_OUT)
-+
-+#if 0
-+static struct pxa2xx_gpio ssp_gpios[3][4] = {
-+	{{ /* SSP1 SND_SOC_DAIFMT_CBM_CFM */
-+		.rx = GPIO26_SSP1RX_MD,
-+		.tx = GPIO25_SSP1TX_MD,
-+		.clk = (23 | GPIO_ALT_FN_2_IN),
-+		.frm = (24 | GPIO_ALT_FN_2_IN),
-+		.sys = GPIO53_SSP1SYSCLK_MD,
-+	},
-+	{ /* SSP1 SND_SOC_DAIFMT_CBS_CFS */
-+		.rx = GPIO26_SSP1RX_MD,
-+		.tx = GPIO25_SSP1TX_MD,
-+		.clk = (23 | GPIO_ALT_FN_2_OUT),
-+		.frm = (24 | GPIO_ALT_FN_2_OUT),
-+		.sys = GPIO53_SSP1SYSCLK_MD,
-+	},
-+	{ /* SSP1 SND_SOC_DAIFMT_CBS_CFM */
-+		.rx = GPIO26_SSP1RX_MD,
-+		.tx = GPIO25_SSP1TX_MD,
-+		.clk = (23 | GPIO_ALT_FN_2_OUT),
-+		.frm = (24 | GPIO_ALT_FN_2_IN),
-+		.sys = GPIO53_SSP1SYSCLK_MD,
-+	},
-+	{ /* SSP1 SND_SOC_DAIFMT_CBM_CFS */
-+		.rx = GPIO26_SSP1RX_MD,
-+		.tx = GPIO25_SSP1TX_MD,
-+		.clk = (23 | GPIO_ALT_FN_2_IN),
-+		.frm = (24 | GPIO_ALT_FN_2_OUT),
-+		.sys = GPIO53_SSP1SYSCLK_MD,
-+	}},
-+	{{ /* SSP2 SND_SOC_DAIFMT_CBM_CFM */
-+		.rx = GPIO11_SSP2RX_MD,
-+		.tx = GPIO13_SSP2TX_MD,
-+		.clk = (22 | GPIO_ALT_FN_3_IN),
-+		.frm = (88 | GPIO_ALT_FN_3_IN),
-+		.sys = GPIO22_SSP2SYSCLK_MD,
-+	},
-+	{ /* SSP2 SND_SOC_DAIFMT_CBS_CFS */
-+		.rx = GPIO11_SSP2RX_MD,
-+		.tx = GPIO13_SSP2TX_MD,
-+		.clk = (22 | GPIO_ALT_FN_3_OUT),
-+		.frm = (88 | GPIO_ALT_FN_3_OUT),
-+		.sys = GPIO22_SSP2SYSCLK_MD,
-+	},
-+	{ /* SSP2 SND_SOC_DAIFMT_CBS_CFM */
-+		.rx = GPIO11_SSP2RX_MD,
-+		.tx = GPIO13_SSP2TX_MD,
-+		.clk = (22 | GPIO_ALT_FN_3_OUT),
-+		.frm = (88 | GPIO_ALT_FN_3_IN),
-+		.sys = GPIO22_SSP2SYSCLK_MD,
-+	},
-+	{ /* SSP2 SND_SOC_DAIFMT_CBM_CFS */
-+		.rx = GPIO11_SSP2RX_MD,
-+		.tx = GPIO13_SSP2TX_MD,
-+		.clk = (22 | GPIO_ALT_FN_3_IN),
-+		.frm = (88 | GPIO_ALT_FN_3_OUT),
-+		.sys = GPIO22_SSP2SYSCLK_MD,
-+	}},
-+	{{ /* SSP3 SND_SOC_DAIFMT_CBM_CFM */
-+		.rx = GPIO82_SSP3RX_MD,
-+		.tx = GPIO81_SSP3TX_MD,
-+		.clk = (84 | GPIO_ALT_FN_3_IN),
-+		.frm = (83 | GPIO_ALT_FN_3_IN),
-+		.sys = GPIO45_SSP3SYSCLK_MD,
-+	},
-+	{ /* SSP3 SND_SOC_DAIFMT_CBS_CFS */
-+		.rx = GPIO82_SSP3RX_MD,
-+		.tx = GPIO81_SSP3TX_MD,
-+		.clk = (84 | GPIO_ALT_FN_3_OUT),
-+		.frm = (83 | GPIO_ALT_FN_3_OUT),
-+		.sys = GPIO45_SSP3SYSCLK_MD,
-+	},
-+	{ /* SSP3 SND_SOC_DAIFMT_CBS_CFM */
-+		.rx = GPIO82_SSP3RX_MD,
-+		.tx = GPIO81_SSP3TX_MD,
-+		.clk = (84 | GPIO_ALT_FN_3_OUT),
-+		.frm = (83 | GPIO_ALT_FN_3_IN),
-+		.sys = GPIO45_SSP3SYSCLK_MD,
-+	},
-+	{ /* SSP3 SND_SOC_DAIFMT_CBM_CFS */
-+		.rx = GPIO82_SSP3RX_MD,
-+		.tx = GPIO81_SSP3TX_MD,
-+		.clk = (84 | GPIO_ALT_FN_3_IN),
-+		.frm = (83 | GPIO_ALT_FN_3_OUT),
-+		.sys = GPIO45_SSP3SYSCLK_MD,
-+	}},
-+};
-+#endif
-+
-+static struct snd_soc_machine mainstone;
-+
-+static int mainstone_hifi_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	unsigned int pll_out = 0, bclk = 0, fmt = 0;
-+	int ret = 0;
-+
-+	/*
-+	 * The WM8753 is far better at generating accurate audio clocks than the
-+	 * pxa2xx I2S controller, so we will use it as master when we can.
-+	 * i.e all rates except 8k and 16k as BCLK must be 64 * rate when the
-+	 * pxa27x or pxa25x is slave. Note this restriction does not apply to SSP
-+	 * I2S emulation mode.
-+	 */
-+	switch (params_rate(params)) {
-+	case 8000:
-+	case 16000:
-+		fmt = SND_SOC_DAIFMT_CBS_CFS;
-+		pll_out = 12288000;
-+		break;
-+	case 48000:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_4;
-+		pll_out = 12288000;
-+		break;
-+	case 96000:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_2;
-+		pll_out = 12288000;
-+		break;
-+	case 11025:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_16;
-+		pll_out = 11289600;
-+		break;
-+	case 22050:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_8;
-+		pll_out = 11289600;
-+		break;
-+	case 44100:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_4;
-+		pll_out = 11289600;
-+		break;
-+	case 88200:
-+		fmt = SND_SOC_DAIFMT_CBM_CFS;
-+		bclk = WM8753_BCLK_DIV_2;
-+		pll_out = 11289600;
-+		break;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai,
-+		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
-+		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the codec system clock for DAC and ADC */
-+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the I2S system clock as input (unused) */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set codec BCLK division for sample rate */
-+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* codec PLL input is 13 MHz */
-+	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 13000000, pll_out);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+static int mainstone_hifi_hw_free(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+
-+	/* disable the PLL */
-+	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
-+}
-+
-+/*
-+ * Mainstone WM8753 HiFi DAI opserations.
-+ */
-+static struct snd_soc_ops mainstone_hifi_ops = {
-+	.hw_params = mainstone_hifi_hw_params,
-+	.hw_free = mainstone_hifi_hw_free,
-+};
-+
-+static int mainstone_voice_startup(struct snd_pcm_substream *substream)
-+{
-+	/* enable USB on the go MUX so we can use SSPFRM2 */
-+	MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL;
-+	MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST;
-+
-+	return 0;
-+}
-+
-+static void mainstone_voice_shutdown(struct snd_pcm_substream *substream)
-+{
-+//	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+
-+	/* disable USB on the go MUX so we can use ttyS0 */
-+	MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL;
-+	MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST;
-+
-+	/* liam may need to tristate DAI */
-+}
-+
-+static int mainstone_voice_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	unsigned int pll_out = 0, bclk = 0, pcmdiv = 0;
-+	int ret = 0;
-+
-+	/*
-+	 * The WM8753 is far better at generating accurate audio clocks than the
-+	 * pxa2xx SSP controller, so we will use it as master when we can.
-+	 */
-+	switch (params_rate(params)) {
-+	case 8000:
-+		pll_out = 12288000;
-+		pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 256kHz */
-+		break;
-+	case 16000:
-+		pll_out = 12288000;
-+		pcmdiv = WM8753_PCM_DIV_3; /* 4.096 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 512kHz */
-+		break;
-+	case 48000:
-+		pll_out = 12288000;
-+		pcmdiv = WM8753_PCM_DIV_1; /* 12.288 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 1.536 MHz */
-+		break;
-+	case 11025:
-+		pll_out = 11289600;
-+		pcmdiv = WM8753_PCM_DIV_4; /* 11.2896 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 352.8 kHz */
-+		break;
-+	case 22050:
-+		pll_out = 11289600;
-+		pcmdiv = WM8753_PCM_DIV_2; /* 11.2896 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 705.6 kHz */
-+		break;
-+	case 44100:
-+		pll_out = 11289600;
-+		pcmdiv = WM8753_PCM_DIV_1; /* 11.2896 MHz */
-+		bclk = WM8753_VXCLK_DIV_8; /* 1.4112 MHz */
-+		break;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the codec system clock for DAC and ADC */
-+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, pll_out,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the SSP system clock as input (unused) */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set codec BCLK division for sample rate */
-+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_VXCLKDIV, bclk);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set codec PCM division for sample rate */
-+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* codec PLL input is 13 MHz */
-+	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 13000000, pll_out);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+static int mainstone_voice_hw_free(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+
-+	/* disable the PLL */
-+	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
-+}
-+
-+static struct snd_soc_ops mainstone_voice_ops = {
-+	.startup = mainstone_voice_startup,
-+	.shutdown = mainstone_voice_shutdown,
-+	.hw_params = mainstone_voice_hw_params,
-+	.hw_free = mainstone_voice_hw_free,
-+};
-+
-+static long mst_audio_suspend_mask;
-+
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	mst_audio_suspend_mask = MST_MSCWR2;
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+/* example machine audio_mapnections */
-+static const char* audio_map[][3] = {
-+
-+	/* mic is connected to mic1 - with bias */
-+	{"MIC1", NULL, "Mic Bias"},
-+	{"MIC1N", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic1 Jack"},
-+	{"Mic Bias", NULL, "Mic1 Jack"},
-+
-+	{"ACIN", NULL, "ACOP"},
-+	{NULL, NULL, NULL},
-+};
-+
-+/* headphone detect support on my board */
-+static const char * hp_pol[] = {"Headphone", "Speaker"};
-+static const struct soc_enum wm8753_enum =
-+	SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol);
-+
-+static const struct snd_kcontrol_new wm8753_mainstone_controls[] = {
-+	SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0),
-+	SOC_ENUM("Headphone Detect Polarity", wm8753_enum),
-+};
-+
-+/*
-+ * This is an example machine initialisation for a wm8753 connected to a
-+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
-+ * to re-route the audio in such an event.
-+ */
-+static int mainstone_wm8753_init(struct snd_soc_codec *codec)
-+{
-+	int i, err;
-+
-+	/* set up mainstone codec pins */
-+	snd_soc_dapm_set_endpoint(codec, "RXP", 0);
-+	snd_soc_dapm_set_endpoint(codec, "RXN", 0);
-+	snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
-+
-+	/* add mainstone specific controls */
-+	for (i = 0; i < ARRAY_SIZE(wm8753_mainstone_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8753_mainstone_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	/* set up mainstone specific audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{ /* Hifi Playback - for similatious use with voice below */
-+	.name = "WM8753",
-+	.stream_name = "WM8753 HiFi",
-+	.cpu_dai = &pxa_i2s_dai,
-+	.codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
-+	.init = mainstone_wm8753_init,
-+	.ops = &mainstone_hifi_ops,
-+},
-+{ /* Voice via BT */
-+	.name = "Bluetooth",
-+	.stream_name = "Voice",
-+	.cpu_dai = &pxa_ssp_dai[1],
-+	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
-+	.ops = &mainstone_voice_ops,
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct wm8753_setup_data mainstone_wm8753_setup = {
-+	.i2c_address = 0x1a,
-+};
-+
-+static struct snd_soc_device mainstone_snd_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm8753,
-+	.codec_data = &mainstone_wm8753_setup,
-+};
-+
-+static struct platform_device *mainstone_snd_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
-+	mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
-+	ret = platform_device_add(mainstone_snd_device);
-+
-+	if (ret)
-+		platform_device_put(mainstone_snd_device);
-+
-+	/* SSP port 2 slave */
-+	pxa_gpio_mode(GPIO11_SSP2RX_MD);
-+	pxa_gpio_mode(GPIO13_SSP2TX_MD);
-+	pxa_gpio_mode(GPIO22_SSP2CLKS_MD);
-+	pxa_gpio_mode(GPIO88_SSP2FRMS_MD);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM8753 Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8974.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_wm8974.c
-@@ -0,0 +1,104 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    30th Oct 2005   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/wm8974.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-i2s.h"
-+
-+static struct snd_soc_machine mainstone;
-+
-+static int mainstone_wm8974_init(struct snd_soc_codec *codec)
-+{
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{
-+	.name = "WM8974",
-+	.stream_name = "WM8974 HiFi",
-+	.cpu_dai = &pxa_i2s_dai,
-+	.codec_dai = &wm8974_dai,
-+	.init = mainstone_wm8974_init,
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct wm8974_setup_data mainstone_wm8974_setup = {
-+	.i2c_address = 0x1a,
-+};
-+
-+static struct snd_soc_device mainstone_snd_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm8974,
-+	.codec_data = &mainstone_wm8974_setup,
-+};
-+
-+static struct platform_device *mainstone_snd_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata);
-+	mainstone_snd_devdata.dev = &mainstone_snd_device->dev;
-+	ret = platform_device_add(mainstone_snd_device);
-+
-+	if (ret)
-+		platform_device_put(mainstone_snd_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_wm9712.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_wm9712.c
-@@ -0,0 +1,172 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    29th Jan 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/wm9712.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-ac97.h"
-+
-+static struct snd_soc_machine mainstone;
-+static long mst_audio_suspend_mask;
-+
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	mst_audio_suspend_mask = MST_MSCWR2;
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+/* mainstone machine dapm widgets */
-+static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = {
-+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
-+};
-+
-+/* example machine interconnections */
-+static const char* intercon[][3] = {
-+
-+	/* mic is connected to mic1 - with bias */
-+	{"MIC1", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic (Internal)"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+/*
-+ * This is an example machine initialisation for a wm8753 connected to a
-+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
-+ * to re-route the audio in such an event.
-+ */
-+static int mainstone_wm9712_init(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	/* set up mainstone codec pins */
-+	snd_soc_dapm_set_endpoint(codec, "RXP", 0);
-+	snd_soc_dapm_set_endpoint(codec, "RXN", 0);
-+	//snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
-+
-+	/* Add mainstone specific widgets */
-+	for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]);
-+	}
-+
-+	/* set up mainstone specific audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{
-+	.name = "AC97",
-+	.stream_name = "AC97 HiFi",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
-+	.init = mainstone_wm9712_init,
-+},
-+{
-+	.name = "AC97 Aux",
-+	.stream_name = "AC97 Aux",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct snd_soc_device mainstone_snd_ac97_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm9712,
-+};
-+
-+static struct platform_device *mainstone_snd_ac97_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_ac97_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
-+	mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
-+
-+	if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
-+		platform_device_put(mainstone_snd_ac97_device);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_ac97_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM9712 Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/mainstone_wm9713.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/mainstone_wm9713.c
-@@ -0,0 +1,318 @@
-+/*
-+ * mainstone.c  --  SoC audio for Mainstone
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c
-+ *  Copyright:	MontaVista Software Inc.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    29th Jan 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/mainstone.h>
-+#include <asm/arch/audio.h>
-+
-+#include "../codecs/wm9713.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-ac97.h"
-+#include "pxa2xx-ssp.h"
-+
-+#define GPIO11_SSP2RX_MD	(11 | GPIO_ALT_FN_2_IN)
-+#define GPIO13_SSP2TX_MD	(13 | GPIO_ALT_FN_1_OUT)
-+#define GPIO22_SSP2CLKS_MD	(22 | GPIO_ALT_FN_3_IN)
-+#define GPIO88_SSP2FRMS_MD	(88 | GPIO_ALT_FN_3_IN)
-+#define GPIO22_SSP2CLKM_MD	(22 | GPIO_ALT_FN_3_OUT)
-+#define GPIO88_SSP2FRMM_MD	(88 | GPIO_ALT_FN_3_OUT)
-+#define GPIO22_SSP2SYSCLK_MD	(22 | GPIO_ALT_FN_2_OUT)
-+
-+static struct snd_soc_machine mainstone;
-+
-+static int mainstone_voice_startup(struct snd_pcm_substream *substream)
-+{
-+	/* enable USB on the go MUX so we can use SSPFRM2 */
-+	MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL;
-+	MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST;
-+	return 0;
-+}
-+
-+static void mainstone_voice_shutdown(struct snd_pcm_substream *substream)
-+{
-+	/* disable USB on the go MUX so we can use ttyS0 */
-+	MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL;
-+	MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST;
-+}
-+
-+static int mainstone_voice_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	unsigned int bclk = 0, pcmdiv = 0;
-+	int ret = 0;
-+
-+	switch (params_rate(params)) {
-+	case 8000:
-+		pcmdiv = WM9713_PCMDIV(12); /* 2.048 MHz */
-+		bclk = WM9713_PCMBCLK_DIV_16; /* 128kHz */
-+		break;
-+	case 16000:
-+		pcmdiv = WM9713_PCMDIV(6); /* 4.096 MHz */
-+		bclk = WM9713_PCMBCLK_DIV_16; /* 256kHz */
-+		break;
-+	case 48000:
-+		pcmdiv = WM9713_PCMDIV(2); /* 12.288 MHz */
-+		bclk = WM9713_PCMBCLK_DIV_16; /* 512kHz */
-+		break;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the SSP system clock as input (unused) */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
-+		SND_SOC_CLOCK_IN);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set codec BCLK division for sample rate */
-+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM9713_PCMBCLK_DIV, bclk);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set codec PCM division for sample rate */
-+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, pcmdiv);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+static struct snd_soc_ops mainstone_voice_ops = {
-+	.startup = mainstone_voice_startup,
-+	.shutdown = mainstone_voice_shutdown,
-+	.hw_params = mainstone_voice_hw_params,
-+};
-+
-+static int test = 0;
-+static int get_test(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = test;
-+	return 0;
-+}
-+
-+static int set_test(struct snd_kcontrol *kcontrol,
-+	struct snd_ctl_elem_value *ucontrol)
-+{
-+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-+
-+	test = ucontrol->value.integer.value[0];
-+	if(test) {
-+
-+	} else {
-+
-+	}
-+	return 0;
-+}
-+
-+static long mst_audio_suspend_mask;
-+
-+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	mst_audio_suspend_mask = MST_MSCWR2;
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_resume(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_probe(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static int mainstone_remove(struct platform_device *pdev)
-+{
-+	MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;
-+	return 0;
-+}
-+
-+static const char* test_function[] = {"Off", "On"};
-+static const struct soc_enum mainstone_enum[] = {
-+	SOC_ENUM_SINGLE_EXT(2, test_function),
-+};
-+
-+static const struct snd_kcontrol_new mainstone_controls[] = {
-+	SOC_ENUM_EXT("ATest Function", mainstone_enum[0], get_test, set_test),
-+};
-+
-+/* mainstone machine dapm widgets */
-+static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = {
-+	SND_SOC_DAPM_MIC("Mic 1", NULL),
-+	SND_SOC_DAPM_MIC("Mic 2", NULL),
-+	SND_SOC_DAPM_MIC("Mic 3", NULL),
-+};
-+
-+/* example machine audio_mapnections */
-+static const char* audio_map[][3] = {
-+
-+	/* mic is connected to mic1 - with bias */
-+	{"MIC1", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic 1"},
-+	/* mic is connected to mic2A - with bias */
-+	{"MIC2A", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic 2"},
-+	/* mic is connected to mic2B - with bias */
-+	{"MIC2B", NULL, "Mic Bias"},
-+	{"Mic Bias", NULL, "Mic 3"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+/*
-+ * This is an example machine initialisation for a wm9713 connected to a
-+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic
-+ * to re-route the audio in such an event.
-+ */
-+static int mainstone_wm9713_init(struct snd_soc_codec *codec)
-+{
-+	int i, err;
-+
-+	/* set up mainstone codec pins */
-+	snd_soc_dapm_set_endpoint(codec, "RXP", 0);
-+	snd_soc_dapm_set_endpoint(codec, "RXN", 0);
-+	//snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
-+
-+	/* Add test specific controls */
-+	for (i = 0; i < ARRAY_SIZE(mainstone_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&mainstone_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	/* Add mainstone specific widgets */
-+	for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]);
-+	}
-+
-+	/* set up mainstone specific audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1],
-+			audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+static struct snd_soc_dai_link mainstone_dai[] = {
-+{
-+	.name = "AC97",
-+	.stream_name = "AC97 HiFi",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
-+	.init = mainstone_wm9713_init,
-+},
-+{
-+	.name = "AC97 Aux",
-+	.stream_name = "AC97 Aux",
-+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
-+},
-+{
-+	.name = "WM9713",
-+	.stream_name = "WM9713 Voice",
-+	.cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
-+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
-+	.ops = &mainstone_voice_ops,
-+},
-+};
-+
-+static struct snd_soc_machine mainstone = {
-+	.name = "Mainstone",
-+	.probe = mainstone_probe,
-+	.remove = mainstone_remove,
-+	.suspend_pre = mainstone_suspend,
-+	.resume_post = mainstone_resume,
-+	.dai_link = mainstone_dai,
-+	.num_links = ARRAY_SIZE(mainstone_dai),
-+};
-+
-+static struct snd_soc_device mainstone_snd_ac97_devdata = {
-+	.machine = &mainstone,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm9713,
-+};
-+
-+static struct platform_device *mainstone_snd_ac97_device;
-+
-+static int __init mainstone_init(void)
-+{
-+	int ret;
-+
-+	mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1);
-+	if (!mainstone_snd_ac97_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata);
-+	mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev;
-+
-+	if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0)
-+		platform_device_put(mainstone_snd_ac97_device);
-+
-+	/* SSP port 2 slave */
-+	pxa_gpio_mode(GPIO11_SSP2RX_MD);
-+	pxa_gpio_mode(GPIO13_SSP2TX_MD);
-+	pxa_gpio_mode(GPIO22_SSP2CLKS_MD);
-+	pxa_gpio_mode(GPIO88_SSP2FRMS_MD);
-+
-+	return ret;
-+}
-+
-+static void __exit mainstone_exit(void)
-+{
-+	platform_device_unregister(mainstone_snd_ac97_device);
-+}
-+
-+module_init(mainstone_init);
-+module_exit(mainstone_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM9713 Mainstone");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/pxa/pxa2xx-ssp.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/pxa2xx-ssp.c
-@@ -0,0 +1,666 @@
-+/*
-+ * pxa2xx-ssp.c  --  ALSA Soc Audio Layer
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    12th Aug 2005   Initial version.
-+ *
-+ * TODO:
-+ *  o Test network mode for > 16bit sample size
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/platform_device.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/initval.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+
-+#include <asm/hardware.h>
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/audio.h>
-+#include <asm/arch/ssp.h>
-+
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-ssp.h"
-+
-+#define PXA_SSP_DEBUG 0
-+
-+#if PXA_SSP_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+
-+/*
-+ * SSP audio private data
-+ */
-+struct ssp_priv {
-+	unsigned int sysclk;
-+};
-+
-+static struct ssp_priv ssp_clk[3];
-+static struct ssp_dev ssp[3];
-+#ifdef CONFIG_PM
-+static struct ssp_state ssp_state[3];
-+#endif
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = {
-+	.name			= "SSP1 PCM Mono out",
-+	.dev_addr		= __PREG(SSDR_P1),
-+	.drcmr			= &DRCMRTXSSDR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = {
-+	.name			= "SSP1 PCM Mono in",
-+	.dev_addr		= __PREG(SSDR_P1),
-+	.drcmr			= &DRCMRRXSSDR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = {
-+	.name			= "SSP1 PCM Stereo out",
-+	.dev_addr		= __PREG(SSDR_P1),
-+	.drcmr			= &DRCMRTXSSDR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = {
-+	.name			= "SSP1 PCM Stereo in",
-+	.dev_addr		= __PREG(SSDR_P1),
-+	.drcmr			= &DRCMRRXSSDR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = {
-+	.name			= "SSP2 PCM Mono out",
-+	.dev_addr		= __PREG(SSDR_P2),
-+	.drcmr			= &DRCMRTXSS2DR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = {
-+	.name			= "SSP2 PCM Mono in",
-+	.dev_addr		= __PREG(SSDR_P2),
-+	.drcmr			= &DRCMRRXSS2DR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = {
-+	.name			= "SSP2 PCM Stereo out",
-+	.dev_addr		= __PREG(SSDR_P2),
-+	.drcmr			= &DRCMRTXSS2DR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = {
-+	.name			= "SSP2 PCM Stereo in",
-+	.dev_addr		= __PREG(SSDR_P2),
-+	.drcmr			= &DRCMRRXSS2DR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = {
-+	.name			= "SSP3 PCM Mono out",
-+	.dev_addr		= __PREG(SSDR_P3),
-+	.drcmr			= &DRCMRTXSS3DR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = {
-+	.name			= "SSP3 PCM Mono in",
-+	.dev_addr		= __PREG(SSDR_P3),
-+	.drcmr			= &DRCMRRXSS3DR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH2,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = {
-+	.name			= "SSP3 PCM Stereo out",
-+	.dev_addr		= __PREG(SSDR_P3),
-+	.drcmr			= &DRCMRTXSS3DR,
-+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = {
-+	.name			= "SSP3 PCM Stereo in",
-+	.dev_addr		= __PREG(SSDR_P3),
-+	.drcmr			= &DRCMRRXSS3DR,
-+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
-+				  DCMD_BURST16 | DCMD_WIDTH4,
-+};
-+
-+static struct pxa2xx_pcm_dma_params *ssp_dma_params[3][4] = {
-+	{&pxa2xx_ssp1_pcm_mono_out, &pxa2xx_ssp1_pcm_mono_in,
-+	&pxa2xx_ssp1_pcm_stereo_out,&pxa2xx_ssp1_pcm_stereo_in,},
-+	{&pxa2xx_ssp2_pcm_mono_out, &pxa2xx_ssp2_pcm_mono_in,
-+	&pxa2xx_ssp2_pcm_stereo_out, &pxa2xx_ssp2_pcm_stereo_in,},
-+	{&pxa2xx_ssp3_pcm_mono_out, &pxa2xx_ssp3_pcm_mono_in,
-+	&pxa2xx_ssp3_pcm_stereo_out,&pxa2xx_ssp3_pcm_stereo_in,},
-+};
-+
-+static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	int ret = 0;
-+
-+	if (!rtd->dai->cpu_dai->active) {
-+		ret = ssp_init (&ssp[cpu_dai->id], cpu_dai->id + 1,
-+			SSP_NO_IRQ);
-+		if (ret < 0)
-+			return ret;
-+		ssp_disable(&ssp[cpu_dai->id]);
-+	}
-+	return ret;
-+}
-+
-+static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+
-+	if (!cpu_dai->active) {
-+		ssp_disable(&ssp[cpu_dai->id]);
-+		ssp_exit(&ssp[cpu_dai->id]);
-+	}
-+}
-+
-+#if defined (CONFIG_PXA27x)
-+static int cken[3] = {CKEN23_SSP1, CKEN3_SSP2, CKEN4_SSP3};
-+#else
-+static int cken[3] = {CKEN3_SSP, CKEN9_NSSP, CKEN10_ASSP};
-+#endif
-+
-+#ifdef CONFIG_PM
-+
-+static int pxa2xx_ssp_suspend(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if (!dai->active)
-+		return 0;
-+
-+	ssp_save_state(&ssp[dai->id], &ssp_state[dai->id]);
-+	pxa_set_cken(cken[dai->id], 0);
-+	return 0;
-+}
-+
-+static int pxa2xx_ssp_resume(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if (!dai->active)
-+		return 0;
-+
-+	pxa_set_cken(cken[dai->id], 1);
-+	ssp_restore_state(&ssp[dai->id], &ssp_state[dai->id]);
-+	ssp_enable(&ssp[dai->id]);
-+
-+	return 0;
-+}
-+
-+#else
-+#define pxa2xx_ssp_suspend	NULL
-+#define pxa2xx_ssp_resume	NULL
-+#endif
-+
-+/*
-+ * Set the SSP ports SYSCLK.
-+ */
-+static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
-+	int clk_id, unsigned int freq, int dir)
-+{
-+	int port = cpu_dai->id + 1;
-+	u32 sscr0 = SSCR0_P(port) &
-+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
-+
-+	dbg("pxa2xx_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d",
-+		cpu_dai->id, clk_id, freq);
-+
-+	switch (clk_id) {
-+	case PXA2XX_SSP_CLK_NET_PLL:
-+		sscr0 |= SSCR0_MOD;
-+	case PXA2XX_SSP_CLK_PLL:
-+		/* Internal PLL is fixed on pxa25x and pxa27x */
-+#ifdef CONFIG_PXA27x
-+		ssp_clk[cpu_dai->id].sysclk = 13000000;
-+#else
-+		ssp_clk[cpu_dai->id].sysclk = 1843200;
-+#endif
-+		break;
-+	case PXA2XX_SSP_CLK_EXT:
-+		ssp_clk[cpu_dai->id].sysclk = freq;
-+		sscr0 |= SSCR0_ECS;
-+		break;
-+	case PXA2XX_SSP_CLK_NET:
-+		ssp_clk[cpu_dai->id].sysclk = freq;
-+		sscr0 |= SSCR0_NCS | SSCR0_MOD;
-+		break;
-+	case PXA2XX_SSP_CLK_AUDIO:
-+		ssp_clk[cpu_dai->id].sysclk = 0;
-+		SSCR0_P(port) |= SSCR0_SerClkDiv(1);
-+		sscr0 |= SSCR0_ADC;
-+		break;
-+	default:
-+		return -ENODEV;
-+	}
-+
-+	/* the SSP CKEN clock must be disabled when changing SSP clock mode */
-+	pxa_set_cken(cken[cpu_dai->id], 0);
-+	SSCR0_P(port) |= sscr0;
-+	pxa_set_cken(cken[cpu_dai->id], 1);
-+	return 0;
-+}
-+
-+/*
-+ * Set the SSP clock dividers.
-+ */
-+static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
-+	int div_id, int div)
-+{
-+	int port = cpu_dai->id + 1;
-+
-+	switch (div_id) {
-+	case PXA2XX_SSP_AUDIO_DIV_ACDS:
-+		SSACD_P(port) &= ~ 0x7;
-+		SSACD_P(port) |= SSACD_ACDS(div);
-+		break;
-+	case PXA2XX_SSP_AUDIO_DIV_SCDB:
-+		SSACD_P(port) &= ~0x8;
-+		if (div == PXA2XX_SSP_CLK_SCDB_1)
-+			SSACD_P(port) |= SSACD_SCDB;
-+		break;
-+	case PXA2XX_SSP_DIV_SCR:
-+		SSCR0_P(port) &= ~SSCR0_SCR;
-+		SSCR0_P(port) |= SSCR0_SerClkDiv(div);
-+		break;
-+	default:
-+		return -ENODEV;
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
-+ */
-+static int pxa2xx_ssp_set_dai_pll(struct snd_soc_cpu_dai *cpu_dai,
-+	int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	int port = cpu_dai->id + 1;
-+
-+	SSACD_P(port) &= ~0x70;
-+	switch (freq_out) {
-+	case 5622000:
-+		break;
-+	case 11345000:
-+		SSACD_P(port) |= (0x1 << 4);
-+		break;
-+	case 12235000:
-+		SSACD_P(port) |= (0x2 << 4);
-+		break;
-+	case 14857000:
-+		SSACD_P(port) |= (0x3 << 4);
-+		break;
-+	case 32842000:
-+		SSACD_P(port) |= (0x4 << 4);
-+		break;
-+	case 48000000:
-+		SSACD_P(port) |= (0x5 << 4);
-+		break;
-+	}
-+	return 0;
-+}
-+
-+/*
-+ * Set the active slots in TDM/Network mode
-+ */
-+static int pxa2xx_ssp_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai,
-+	unsigned int mask, int slots)
-+{
-+	int port = cpu_dai->id + 1;
-+
-+	SSCR0_P(port) &= ~SSCR0_SlotsPerFrm(7);
-+
-+	/* set number of active slots */
-+	SSCR0_P(port) |= SSCR0_SlotsPerFrm(slots);
-+
-+	/* set active slot mask */
-+	SSTSA_P(port) = mask;
-+	SSRSA_P(port) = mask;
-+	return 0;
-+}
-+
-+/*
-+ * Tristate the SSP DAI lines
-+ */
-+static int pxa2xx_ssp_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai,
-+	int tristate)
-+{
-+	int port = cpu_dai->id + 1;
-+
-+	if (tristate)
-+		SSCR1_P(port) &= ~SSCR1_TTE;
-+	else
-+		SSCR1_P(port) |= SSCR1_TTE;
-+
-+	return 0;
-+}
-+
-+/*
-+ * Set up the SSP DAI format.
-+ * The SSP Port must be inactive before calling this function as the
-+ * physical interface format is changed.
-+ */
-+static int pxa2xx_ssp_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
-+		unsigned int fmt)
-+{
-+	int port = cpu_dai->id + 1;
-+
-+	/* reset port settings */
-+	SSCR0_P(port) = 0;
-+	SSCR1_P(port) = 0;
-+	SSPSP_P(port) = 0;
-+
-+	/* NOTE: I2S emulation is still very much work in progress here */
-+
-+	/* FIXME: this is what wince uses for msb */
-+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB) {
-+		SSCR0_P(port) = SSCR0_EDSS | SSCR0_TISSP | SSCR0_DataSize(16);
-+		goto master;
-+	}
-+
-+	/* check for I2S emulation mode - handle it separately  */
-+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) {
-+		/* 8.4.11 */
-+
-+		/* Only SSCR0[NCS] or SSCR0[ECS] bit fields settings are optional */
-+		SSCR0_P(port) = SSCR0_EDSS | SSCR0_PSP | SSCR0_DataSize(16);
-+
-+		/* set FIFO thresholds */
-+		SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1);
-+
-+		/* normal: */
-+		/* all bit fields must be cleared except: FSRT = 1 and
-+		 * SFRMWDTH = 16, DMYSTART=0,1) */
-+		SSPSP_P(port) = SSPSP_FSRT | SSPSP_SFRMWDTH(16) | SSPSP_DMYSTRT(0);
-+		goto master;
-+	}
-+
-+	SSCR0_P(port) |= SSCR0_PSP;
-+	SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) |
-+		SSCR1_TRAIL | SSCR1_RWOT;
-+
-+master:
-+	switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		SSCR1_P(port) |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR);
-+		break;
-+	case SND_SOC_DAIFMT_CBM_CFS:
-+		SSCR1_P(port) |= SSCR1_SCLKDIR;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFM:
-+		SSCR1_P(port) |= SSCR1_SFRMDIR;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		SSPSP_P(port) |= SSPSP_SFRMP | SSPSP_FSRT;
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_DSP_A:
-+		SSPSP_P(port) |= SSPSP_DMYSTRT(1);
-+	case SND_SOC_DAIFMT_DSP_B:
-+		SSPSP_P(port) |= SSPSP_SCMODE(2);
-+		break;
-+	case SND_SOC_DAIFMT_I2S:
-+	case SND_SOC_DAIFMT_MSB:
-+		/* handled above */
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * Set the SSP audio DMA parameters and sample size.
-+ * Can be called multiple times by oss emulation.
-+ */
-+static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	int dma = 0, chn = params_channels(params);
-+	int port = cpu_dai->id + 1;
-+
-+	/* select correct DMA params */
-+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
-+		dma = 1; /* capture DMA offset is 1,3 */
-+	if (chn == 2)
-+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
-+	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
-+
-+	dbg("pxa2xx_ssp_hw_params: dma %d", dma);
-+
-+	/* we can only change the settings if the port is not in use */
-+	if (SSCR0_P(port) & SSCR0_SSE)
-+		return 0;
-+
-+	/* clear selected SSP bits */
-+	SSCR0_P(port) &= ~(SSCR0_DSS | SSCR0_EDSS);
-+
-+	/* bit size */
-+	switch(params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		SSCR0_P(port) |= SSCR0_DataSize(16);
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		SSCR0_P(port) |=(SSCR0_EDSS | SSCR0_DataSize(8));
-+		/* we must be in network mode (2 slots) for 24 bit stereo */
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		SSCR0_P(port) |= (SSCR0_EDSS | SSCR0_DataSize(16));
-+		/* we must be in network mode (2 slots) for 32 bit stereo */
-+		break;
-+	}
-+
-+	dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x",
-+		SSCR0_P(port), SSCR1_P(port),
-+		SSTO_P(port), SSPSP_P(port),
-+		SSSR_P(port), SSACD_P(port));
-+
-+	return 0;
-+}
-+
-+static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	int ret = 0;
-+	int port = cpu_dai->id + 1;
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+		ssp_enable(&ssp[cpu_dai->id]);
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			SSCR1_P(port) |= SSCR1_TSRE;
-+		else
-+			SSCR1_P(port) |= SSCR1_RSRE;
-+		SSSR_P(port) |= SSSR_P(port);
-+		break;
-+	case SNDRV_PCM_TRIGGER_START:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			SSCR1_P(port) |= SSCR1_TSRE;
-+		else
-+			SSCR1_P(port) |= SSCR1_RSRE;
-+		ssp_enable(&ssp[cpu_dai->id]);
-+		break;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			SSCR1_P(port) &= ~SSCR1_TSRE;
-+		else
-+			SSCR1_P(port) &= ~SSCR1_RSRE;
-+		break;
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+		ssp_disable(&ssp[cpu_dai->id]);
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			SSCR1_P(port) &= ~SSCR1_TSRE;
-+		else
-+			SSCR1_P(port) &= ~SSCR1_RSRE;
-+		break;
-+
-+	default:
-+		ret = -EINVAL;
-+	}
-+
-+	dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x",
-+		SSCR0_P(port), SSCR1_P(port),
-+		SSTO_P(port), SSPSP_P(port),
-+		SSSR_P(port));
-+
-+	return ret;
-+}
-+
-+#define PXA2XX_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-+
-+struct snd_soc_cpu_dai pxa_ssp_dai[] = {
-+	{	.name = "pxa2xx-ssp1",
-+		.id = 0,
-+		.type = SND_SOC_DAI_PCM,
-+		.suspend = pxa2xx_ssp_suspend,
-+		.resume = pxa2xx_ssp_resume,
-+		.playback = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.capture = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.ops = {
-+			.startup = pxa2xx_ssp_startup,
-+			.shutdown = pxa2xx_ssp_shutdown,
-+			.trigger = pxa2xx_ssp_trigger,
-+			.hw_params = pxa2xx_ssp_hw_params,},
-+		.dai_ops = {
-+			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
-+			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
-+			.set_pll = pxa2xx_ssp_set_dai_pll,
-+			.set_fmt = pxa2xx_ssp_set_dai_fmt,
-+			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
-+			.set_tristate = pxa2xx_ssp_set_dai_tristate,
-+		},
-+	},
-+	{	.name = "pxa2xx-ssp2",
-+		.id = 1,
-+		.type = SND_SOC_DAI_PCM,
-+		.suspend = pxa2xx_ssp_suspend,
-+		.resume = pxa2xx_ssp_resume,
-+		.playback = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.capture = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.ops = {
-+			.startup = pxa2xx_ssp_startup,
-+			.shutdown = pxa2xx_ssp_shutdown,
-+			.trigger = pxa2xx_ssp_trigger,
-+			.hw_params = pxa2xx_ssp_hw_params,},
-+		.dai_ops = {
-+			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
-+			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
-+			.set_pll = pxa2xx_ssp_set_dai_pll,
-+			.set_fmt = pxa2xx_ssp_set_dai_fmt,
-+			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
-+			.set_tristate = pxa2xx_ssp_set_dai_tristate,
-+		},
-+	},
-+	{	.name = "pxa2xx-ssp3",
-+		.id = 2,
-+		.type = SND_SOC_DAI_PCM,
-+		.suspend = pxa2xx_ssp_suspend,
-+		.resume = pxa2xx_ssp_resume,
-+		.playback = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.capture = {
-+			.channels_min = 1,
-+			.channels_max = 2,
-+			.rates = PXA2XX_SSP_RATES,
-+			.formats = PXA2XX_SSP_FORMATS,},
-+		.ops = {
-+			.startup = pxa2xx_ssp_startup,
-+			.shutdown = pxa2xx_ssp_shutdown,
-+			.trigger = pxa2xx_ssp_trigger,
-+			.hw_params = pxa2xx_ssp_hw_params,},
-+		.dai_ops = {
-+			.set_sysclk = pxa2xx_ssp_set_dai_sysclk,
-+			.set_clkdiv = pxa2xx_ssp_set_dai_clkdiv,
-+			.set_pll = pxa2xx_ssp_set_dai_pll,
-+			.set_fmt = pxa2xx_ssp_set_dai_fmt,
-+			.set_tdm_slot = pxa2xx_ssp_set_dai_tdm_slot,
-+			.set_tristate = pxa2xx_ssp_set_dai_tristate,
-+		},
-+	},
-+};
-+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("pxa2xx SSP/PCM SoC Interface");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/imx/imx-ssi.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx-ssi.c
-@@ -0,0 +1,591 @@
-+/*
-+ * imx-ssi.c  --  SSI driver for Freescale IMX
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Based on mxc-alsa-mc13783 (C) 2006 Freescale.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    29th Aug 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/platform_device.h>
-+#include <linux/slab.h>
-+#include <linux/dma-mapping.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <asm/arch/dma.h>
-+#include <asm/arch/spba.h>
-+#include <asm/arch/clock.h>
-+#include <asm/mach-types.h>
-+#include <asm/hardware.h>
-+
-+#include "imx-ssi.h"
-+#include "imx31-pcm.h"
-+
-+static struct mxc_pcm_dma_params imx_ssi1_pcm_stereo_out = {
-+	.name			= "SSI1 PCM Stereo out",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = emi_2_per,
-+		.watermark_level = SDMA_TXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI1_STX0,
-+		.event_id = DMA_REQ_SSI1_TX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static struct mxc_pcm_dma_params imx_ssi1_pcm_stereo_in = {
-+	.name			= "SSI1 PCM Stereo in",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_RXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI1_SRX0,
-+		.event_id = DMA_REQ_SSI1_RX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static struct mxc_pcm_dma_params imx_ssi2_pcm_stereo_out = {
-+	.name			= "SSI2 PCM Stereo out",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_TXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI2_STX0,
-+		.event_id = DMA_REQ_SSI2_TX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static struct mxc_pcm_dma_params imx_ssi2_pcm_stereo_in = {
-+	.name			= "SSI2 PCM Stereo in",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_RXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI2_SRX0,
-+		.event_id = DMA_REQ_SSI2_RX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+/*
-+ * SSI system clock configuration.
-+ */
-+static int imx_ssi_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
-+	int clk_id, unsigned int freq, int dir)
-+{
-+	u32 scr;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1)
-+		scr = __raw_readw(SSI1_SCR);
-+	else
-+		scr = __raw_readw(SSI2_SCR);
-+
-+	switch (clk_id) {
-+	case IMX_SSP_SYS_CLK:
-+		if (dir == SND_SOC_CLOCK_OUT)
-+			scr |= SSI_SCR_SYS_CLK_EN;
-+		else
-+			scr &= ~SSI_SCR_SYS_CLK_EN;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1)
-+		__raw_writew(scr, SSI1_SCR);
-+	else
-+		__raw_writew(scr, SSI2_SCR);
-+
-+	return 0;
-+}
-+
-+/*
-+ * SSI Clock dividers
-+ */
-+static int imx_ssi_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
-+	int div_id, int div)
-+{
-+	u32 stccr;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1)
-+		stccr = __raw_readw(SSI1_STCCR);
-+	else
-+		stccr = __raw_readw(SSI2_STCCR);
-+
-+	switch (div_id) {
-+	case IMX_SSI_DIV_2:
-+		stccr &= ~SSI_STCCR_DIV2;
-+		stccr |= div;
-+		break;
-+	case IMX_SSI_DIV_PSR:
-+		stccr &= ~SSI_STCCR_PSR;
-+		stccr |= div;
-+		break;
-+	case IMX_SSI_DIV_PM:
-+		stccr &= ~0xff;
-+		stccr |= SSI_STCCR_PM(div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1)
-+		__raw_writew(stccr, SSI1_STCCR);
-+	else
-+		__raw_writew(stccr, SSI2_STCCR);
-+
-+	return 0;
-+}
-+
-+/*
-+ * SSI Network Mode or TDM slots configuration.
-+ */
-+static int imx_ssi_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai,
-+	unsigned int mask, int slots)
-+{
-+	u32 stmsk, srmsk, scr, stccr;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		stmsk = __raw_readw(SSI1_STMSK);
-+		srmsk = __raw_readw(SSI1_SRMSK);
-+		scr = __raw_readw(SSI1_SCR);
-+		stccr = __raw_readw(SSI1_STCCR);
-+	} else {
-+		stmsk = __raw_readw(SSI2_STMSK);
-+		srmsk = __raw_readw(SSI2_SRMSK);
-+		scr = __raw_readw(SSI2_SCR);
-+		stccr = __raw_readw(SSI2_STCCR);
-+	}
-+
-+	stmsk = srmsk = mask;
-+	scr |= SSI_SCR_NET;
-+	stccr &= ~0x1f00;
-+	stccr |= SSI_STCCR_DC(slots);
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		__raw_writew(stmsk, SSI1_STMSK);
-+		__raw_writew(srmsk, SSI1_SRMSK);
-+		__raw_writew(scr, SSI1_SCR);
-+		__raw_writew(stccr, SSI1_STCCR);
-+	} else {
-+		__raw_writew(stmsk, SSI2_STMSK);
-+		__raw_writew(srmsk, SSI2_SRMSK);
-+		__raw_writew(scr, SSI2_SCR);
-+		__raw_writew(stccr, SSI2_STCCR);
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * SSI DAI format configuration.
-+ */
-+static int imx_ssi_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
-+		unsigned int fmt)
-+{
-+	u32 stcr = 0, srcr = 0;
-+
-+	/* DAI mode */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		stcr |= SSI_STCR_TSCKP | SSI_STCR_TFSI |
-+			SSI_STCR_TEFS | SSI_STCR_TXBIT0;
-+		srcr |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI |
-+			SSI_SRCR_REFS | SSI_SRCR_RXBIT0;
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		stcr |= SSI_STCR_TSCKP | SSI_STCR_TFSI | SSI_STCR_TXBIT0;
-+		srcr |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | SSI_SRCR_RXBIT0;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		stcr |= SSI_STCR_TEFS; // data 1 bit after sync
-+		srcr |= SSI_SRCR_REFS; // data 1 bit after sync
-+	case SND_SOC_DAIFMT_DSP_A:
-+		stcr |= SSI_STCR_TFSL; // frame is 1 bclk long
-+		srcr |= SSI_SRCR_RFSL; // frame is 1 bclk long
-+
-+		/* DAI clock inversion */
-+		switch(fmt & SND_SOC_DAIFMT_INV_MASK) {
-+		case SND_SOC_DAIFMT_IB_IF:
-+			stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
-+			srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP;
-+			break;
-+		case SND_SOC_DAIFMT_IB_NF:
-+			stcr |= SSI_STCR_TSCKP;
-+			srcr |= SSI_SRCR_RSCKP;
-+			break;
-+		case SND_SOC_DAIFMT_NB_IF:
-+			stcr |= SSI_STCR_TFSI;
-+			srcr |= SSI_SRCR_RFSI;
-+			break;
-+		}
-+		break;
-+	}
-+
-+	/* DAI clock master masks */
-+	switch(fmt & SND_SOC_DAIFMT_CLOCK_MASK){
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR;
-+		srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFM:
-+		stcr |= SSI_STCR_TFDIR;
-+		srcr |= SSI_SRCR_RFDIR;
-+		break;
-+	case SND_SOC_DAIFMT_CBM_CFS:
-+		stcr |= SSI_STCR_TXDIR;
-+		srcr |= SSI_SRCR_RXDIR;
-+		break;
-+	}
-+
-+	/* async */
-+	//if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC)
-+	//	SSI1_SCR |= SSI_SCR_SYN;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		__raw_writew(stcr, SSI1_STCR);
-+		__raw_writew(0, SSI1_STCCR);
-+		__raw_writew(srcr, SSI1_SRCR);
-+		__raw_writew(0, SSI1_SRCCR);
-+	} else {
-+		__raw_writew(stcr, SSI2_STCR);
-+		__raw_writew(0, SSI2_STCCR);
-+		__raw_writew(srcr, SSI2_SRCR);
-+		__raw_writew(0, SSI2_SRCCR);
-+	}
-+
-+	return 0;
-+}
-+
-+static int imx_ssi_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai,
-+	int tristate)
-+{
-+	// via GPIO ??
-+	return 0;
-+}
-+
-+static int imx_ssi_startup(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1)
-+		mxc_clks_enable(SSI1_BAUD);
-+	else
-+		mxc_clks_enable(SSI2_BAUD);
-+	return 0;
-+}
-+
-+static int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	u32 stccr, stcr;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		stccr = __raw_readw(SSI1_STCCR) & 0x600ff;
-+		stcr = __raw_readw(SSI1_STCR);
-+	} else {
-+		stccr = __raw_readw(SSI2_STCCR) & 0x600ff;
-+		stcr = __raw_readw(SSI2_STCR);
-+	}
-+
-+	/* DAI data (word) size */
-+	switch(params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		stccr |= SSI_STCCR_WL(16);
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		stccr |= SSI_STCCR_WL(20);
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		stccr |= SSI_STCCR_WL(24);
-+		break;
-+	}
-+
-+	/* TDM - todo, only fifo 0 atm */
-+	stcr |= SSI_STCR_TFEN0;
-+	stccr |= SSI_STCCR_DC(params_channels(params));
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		__raw_writew(stcr, SSI1_STCR);
-+		__raw_writew(stccr, SSI1_STCCR);
-+	} else {
-+		__raw_writew(stcr, SSI2_STCR);
-+		__raw_writew(stccr, SSI2_STCCR);
-+	}
-+
-+	return 0;
-+}
-+
-+static int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	u32 srccr, srcr;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		srccr = __raw_readw(SSI1_SRCCR) & 0x600ff;
-+		srcr = __raw_readw(SSI1_SRCR);
-+	} else {
-+		srccr = __raw_readw(SSI2_SRCCR) & 0x600ff;
-+		srcr = __raw_readw(SSI2_SRCR);
-+	}
-+
-+	/* DAI data (word) size */
-+	switch(params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		srccr |= SSI_SRCCR_WL(16);
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		srccr |= SSI_SRCCR_WL(20);
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		srccr |= SSI_SRCCR_WL(24);
-+		break;
-+	}
-+
-+	/* TDM - todo, only fifo 0 atm */
-+	srcr |= SSI_SRCR_RFEN0;
-+	srccr |= SSI_SRCCR_DC(params_channels(params));
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		__raw_writew(srcr, SSI1_SRCR);
-+		__raw_writew(srccr, SSI1_SRCCR);
-+	} else {
-+		__raw_writew(srcr, SSI2_SRCR);
-+		__raw_writew(srccr, SSI2_SRCCR);
-+	}
-+	return 0;
-+}
-+
-+static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+
-+	/* Tx/Rx config */
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		if (cpu_dai->id == IMX_DAI_SSI1)
-+			cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out;
-+		else
-+			cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out;
-+		return imx_ssi_hw_tx_params(substream, params);
-+	} else {
-+		if (cpu_dai->id == IMX_DAI_SSI1)
-+			cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in;
-+		else
-+			cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in;
-+		return imx_ssi_hw_rx_params(substream, params);
-+	}
-+}
-+
-+static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	u32 scr, sier;
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		scr = __raw_readw(SSI1_SCR) & 0x600ff;
-+		sier = __raw_readw(SSI1_SIER);
-+	} else {
-+		scr = __raw_readw(SSI2_SCR) & 0x600ff;
-+		sier = __raw_readw(SSI2_SIER);
-+	}
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+			scr |= SSI_SCR_TE;
-+			sier |= SSI_SIER_TDMAE;
-+		} else {
-+			scr |= SSI_SCR_RE;
-+			sier |= SSI_SIER_RDMAE;
-+		}
-+		scr |= SSI_SCR_SSIEN;
-+		break;
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			scr |= SSI_SCR_TE;
-+		else
-+			scr |= SSI_SCR_RE;
-+		scr |= SSI_SCR_SSIEN;
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			sier |= SSI_SIER_TDMAE;
-+		else
-+			sier |= SSI_SIER_RDMAE;
-+	break;
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+		scr &= ~SSI_SCR_SSIEN;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			scr &= ~SSI_SCR_TE;
-+		else
-+			scr &= ~SSI_SCR_RE;
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+			sier &= ~SSI_SIER_TDMAE;
-+		else
-+			sier &= ~SSI_SIER_TDMAE;
-+	break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	if (cpu_dai->id == IMX_DAI_SSI1) {
-+		__raw_writew(scr, SSI1_SCR);
-+		__raw_writew(sier, SSI1_SIER);
-+	} else {
-+		__raw_writew(scr, SSI2_SCR);
-+		__raw_writew(sier, SSI2_SIER);
-+	}
-+
-+	return 0;
-+}
-+
-+static void imx_ssi_shutdown(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+
-+	/* shutdown SSI */
-+	if (!cpu_dai->active) {
-+		if(cpu_dai->id == IMX_DAI_SSI1) {
-+			__raw_writew(__raw_readw(SSI1_SCR) & ~SSI_SCR_SSIEN, SSI1_SCR);
-+			mxc_clks_disable(SSI1_BAUD);
-+		} else {
-+			__raw_writew(__raw_readw(SSI2_SCR) & ~SSI_SCR_SSIEN, SSI2_SCR);
-+			mxc_clks_disable(SSI2_BAUD);
-+		}
-+	}
-+}
-+
-+#ifdef CONFIG_PM
-+static int imx_ssi_suspend(struct platform_device *dev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if(!dai->active)
-+		return 0;
-+
-+	// do we need to disable any clocks
-+
-+	return 0;
-+}
-+
-+static int imx_ssi_resume(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if(!dai->active)
-+		return 0;
-+
-+	// do we need to enable any clocks
-+	return 0;
-+}
-+
-+#else
-+#define imx_ssi_suspend	NULL
-+#define imx_ssi_resume	NULL
-+#endif
-+
-+#define IMX_SSI_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
-+	SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
-+	SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
-+	SNDRV_PCM_RATE_96000)
-+
-+#define IMX_SSI_BITS \
-+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_cpu_dai imx_ssi_pcm_dai[] = {
-+{
-+	.name = "imx-i2s-1",
-+	.id = IMX_DAI_SSI1,
-+	.type = SND_SOC_DAI_I2S,
-+	.suspend = imx_ssi_suspend,
-+	.resume = imx_ssi_resume,
-+	.playback = {
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.formats = IMX_SSI_BITS,
-+		.rates = IMX_SSI_RATES,},
-+	.capture = {
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.formats = IMX_SSI_BITS,
-+		.rates = IMX_SSI_RATES,},
-+	.ops = {
-+		.startup = imx_ssi_startup,
-+		.shutdown = imx_ssi_shutdown,
-+		.trigger = imx_ssi_trigger,
-+		.hw_params = imx_ssi_hw_params,},
-+	.dai_ops = {
-+		.set_sysclk = imx_ssi_set_dai_sysclk,
-+		.set_clkdiv = imx_ssi_set_dai_clkdiv,
-+		.set_fmt = imx_ssi_set_dai_fmt,
-+		.set_tdm_slot = imx_ssi_set_dai_tdm_slot,
-+		.set_tristate = imx_ssi_set_dai_tristate,
-+	},
-+},
-+{
-+	.name = "imx-i2s-2",
-+	.id = IMX_DAI_SSI2,
-+	.type = SND_SOC_DAI_I2S,
-+	.suspend = imx_ssi_suspend,
-+	.resume = imx_ssi_resume,
-+	.playback = {
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.formats = IMX_SSI_BITS,
-+		.rates = IMX_SSI_RATES,},
-+	.capture = {
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.formats = IMX_SSI_BITS,
-+		.rates = IMX_SSI_RATES,},
-+	.ops = {
-+		.startup = imx_ssi_startup,
-+		.shutdown = imx_ssi_shutdown,
-+		.trigger = imx_ssi_trigger,
-+		.hw_params = imx_ssi_hw_params,},
-+	.dai_ops = {
-+		.set_sysclk = imx_ssi_set_dai_sysclk,
-+		.set_clkdiv = imx_ssi_set_dai_clkdiv,
-+		.set_fmt = imx_ssi_set_dai_fmt,
-+		.set_tdm_slot = imx_ssi_set_dai_tdm_slot,
-+		.set_tristate = imx_ssi_set_dai_tristate,
-+	},
-+},};
-+EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("i.MX ASoC I2S driver");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/imx/Kconfig
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/Kconfig
-@@ -0,0 +1,31 @@
-+menu "SoC Audio for the Freescale i.MX"
-+
-+config SND_MXC_SOC
-+	tristate "SoC Audio for the Freescale i.MX CPU"
-+	depends on ARCH_MXC && SND
-+	select SND_PCM
-+	help
-+	  Say Y or M if you want to add support for codecs attached to
-+	  the MXC AC97, I2S or SSP interface. You will also need
-+	  to select the audio interfaces to support below.
-+
-+config SND_MXC_AC97
-+	tristate
-+	select SND_AC97_CODEC
-+
-+config SND_MXC_SOC_AC97
-+	tristate
-+	select AC97_BUS
-+
-+config SND_MXC_SOC_SSI
-+	tristate
-+
-+config SND_SOC_MX31ADS_WM8753
-+	tristate "SoC Audio support for MX31 - WM8753"
-+	depends on SND_MXC_SOC && ARCH_MX3
-+	select SND_MXC_SOC_SSI
-+	help
-+	  Say Y if you want to add support for SoC audio on MX31ADS
-+	  with the WM8753.
-+
-+endmenu
-Index: linux-2.6.21-moko/sound/soc/imx/Makefile
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/Makefile
-@@ -0,0 +1,18 @@
-+# i.MX Platform Support
-+snd-soc-imx21-objs := imx21-pcm.o
-+snd-soc-imx31-objs := imx31-pcm.o
-+snd-soc-imx-ac97-objs := imx-ac97.o
-+snd-soc-imx-ssi-objs := imx-ssi.o
-+
-+obj-$(CONFIG_SND_MXC_SOC) += snd-soc-imx31.o
-+obj-$(CONFIG_SND_MXC_SOC_AC97) += snd-soc-imx-ac97.o
-+obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-imx-ssi.o
-+
-+# i.MX Machine Support
-+snd-soc-mx31ads-wm8753-objs := mx31ads_wm8753.o
-+obj-$(CONFIG_SND_SOC_MX31ADS_WM8753) += snd-soc-mx31ads-wm8753.o
-+snd-soc-mx21ads-wm8753-objs := mx21ads_wm8753.o
-+obj-$(CONFIG_SND_SOC_MX21ADS_WM8753) += snd-soc-mx21ads-wm8753.o
-+snd-soc-mx21ads-wm8731-objs := mx21ads_wm8731.o
-+obj-$(CONFIG_SND_SOC_MX21ADS_WM8731) += snd-soc-mx21ads-wm8731.o
-+
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8711.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8711.c
-@@ -0,0 +1,715 @@
-+/*
-+ * wm8711.c  --  WM8711 ALSA SoC Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics
-+ *
-+ * Author: Mike Arthur <linux at wolfsonmicro.com>
-+ *
-+ * Based on wm8731.c by Richard Purdie
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8711.h"
-+
-+#define AUDIO_NAME "wm8711"
-+#define WM8711_VERSION "0.3"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8711_DEBUG 0
-+
-+#ifdef WM8711_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8711;
-+
-+/* codec private data */
-+struct wm8711_priv {
-+	unsigned int sysclk;
-+};
-+
-+/*
-+ * wm8711 register cache
-+ * We can't read the WM8711 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ * There is no point in caching the reset register
-+ */
-+static const u16 wm8711_reg[WM8711_CACHEREGNUM] = {
-+    0x0079, 0x0079, 0x000a, 0x0008,
-+    0x009f, 0x000a, 0x0000, 0x0000
-+};
-+
-+/*
-+ * read wm8711 register cache
-+ */
-+static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec * codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8711_RESET)
-+		return 0;
-+	if (reg >= WM8711_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8711 register cache
-+ */
-+static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8711_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8711 register space
-+ */
-+static int wm8711_write(struct snd_soc_codec * codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8753 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8711_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8711_reset(c)	wm8711_write(c, WM8711_RESET, 0)
-+
-+static const struct snd_kcontrol_new wm8711_snd_controls[] = {
-+
-+SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V,
-+	0, 127, 0),
-+SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V,
-+	7, 1, 0),
-+
-+};
-+
-+/* add non dapm controls */
-+static int wm8711_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8711_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Output Mixer */
-+static const snd_kcontrol_new_t wm8711_output_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0),
-+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0),
-+};
-+
-+static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = {
-+SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1,
-+	&wm8711_output_mixer_controls[0],
-+	ARRAY_SIZE(wm8711_output_mixer_controls)),
-+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1),
-+SND_SOC_DAPM_OUTPUT("LOUT"),
-+SND_SOC_DAPM_OUTPUT("LHPOUT"),
-+SND_SOC_DAPM_OUTPUT("ROUT"),
-+SND_SOC_DAPM_OUTPUT("RHPOUT"),
-+};
-+
-+static const char *intercon[][3] = {
-+	/* output mixer */
-+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
-+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
-+
-+	/* outputs */
-+	{"RHPOUT", NULL, "Output Mixer"},
-+	{"ROUT", NULL, "Output Mixer"},
-+	{"LHPOUT", NULL, "Output Mixer"},
-+	{"LOUT", NULL, "Output Mixer"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8711_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8711_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8711_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1],
-+			intercon[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct _coeff_div {
-+	u32 mclk;
-+	u32 rate;
-+	u16 fs;
-+	u8 sr:4;
-+	u8 bosr:1;
-+	u8 usb:1;
-+};
-+
-+/* codec mclk clock divider coefficients */
-+static const struct _coeff_div coeff_div[] = {
-+	/* 48k */
-+	{12288000, 48000, 256, 0x0, 0x0, 0x0},
-+	{18432000, 48000, 384, 0x0, 0x1, 0x0},
-+	{12000000, 48000, 250, 0x0, 0x0, 0x1},
-+
-+	/* 32k */
-+	{12288000, 32000, 384, 0x6, 0x0, 0x0},
-+	{18432000, 32000, 576, 0x6, 0x1, 0x0},
-+	{12000000, 32000, 375, 0x6, 0x0, 0x1},
-+
-+	/* 8k */
-+	{12288000, 8000, 1536, 0x3, 0x0, 0x0},
-+	{18432000, 8000, 2304, 0x3, 0x1, 0x0},
-+	{11289600, 8000, 1408, 0xb, 0x0, 0x0},
-+	{16934400, 8000, 2112, 0xb, 0x1, 0x0},
-+	{12000000, 8000, 1500, 0x3, 0x0, 0x1},
-+
-+	/* 96k */
-+	{12288000, 96000, 128, 0x7, 0x0, 0x0},
-+	{18432000, 96000, 192, 0x7, 0x1, 0x0},
-+	{12000000, 96000, 125, 0x7, 0x0, 0x1},
-+
-+	/* 44.1k */
-+	{11289600, 44100, 256, 0x8, 0x0, 0x0},
-+	{16934400, 44100, 384, 0x8, 0x1, 0x0},
-+	{12000000, 44100, 272, 0x8, 0x1, 0x1},
-+
-+	/* 88.2k */
-+	{11289600, 88200, 128, 0xf, 0x0, 0x0},
-+	{16934400, 88200, 192, 0xf, 0x1, 0x0},
-+	{12000000, 88200, 136, 0xf, 0x1, 0x1},
-+};
-+
-+static inline int get_coeff(int mclk, int rate)
-+{
-+	int i;
-+
-+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
-+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
-+			return i;
-+	}
-+	return 0;
-+}
-+
-+static int wm8711_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct wm8711_priv *wm8711 = codec->private_data;
-+	u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc;
-+	int i = get_coeff(wm8711->sysclk, params_rate(params));
-+	u16 srate = (coeff_div[i].sr << 2) |
-+		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
-+
-+	wm8711_write(codec, WM8711_SRATE, srate);
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0008;
-+		break;
-+	}
-+
-+	wm8711_write(codec, WM8711_IFACE, iface);
-+	return 0;
-+}
-+
-+static int wm8711_pcm_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+    /* set active */
-+    wm8711_write(codec, WM8711_ACTIVE, 0x0001);
-+    return 0;
-+}
-+
-+static void wm8711_shutdown(struct snd_pcm_substream *substream)
-+{
-+    struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+    struct snd_soc_device *socdev = rtd->socdev;
-+    struct snd_soc_codec *codec = socdev->codec;
-+
-+    /* deactivate */
-+    if (!codec->active) {
-+        udelay(50);
-+        wm8711_write(codec, WM8711_ACTIVE, 0x0);
-+    }
-+}
-+
-+static int wm8711_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+    u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7;
-+
-+    if (mute)
-+        wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8);
-+    else
-+        wm8711_write(codec, WM8711_APDIGI, mute_reg);
-+
-+	return 0;
-+}
-+
-+static int wm8711_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
-+		int clk_id, unsigned int freq, int dir)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	struct wm8711_priv *wm8711 = codec->private_data;
-+
-+	switch (freq) {
-+	case 11289600:
-+	case 12000000:
-+	case 12288000:
-+	case 16934400:
-+	case 18432000:
-+		wm8711->sysclk = freq;
-+		return 0;
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8711_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = 0;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		iface |= 0x0040;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		iface |= 0x0013;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0090;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0080;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0010;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* set iface */
-+	wm8711_write(codec, WM8711_IFACE, iface);
-+	return 0;
-+}
-+
-+
-+static int wm8711_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f;
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, osc on, dac unmute */
-+		wm8711_write(codec, WM8711_PWR, reg);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, */
-+		wm8711_write(codec, WM8711_PWR, reg | 0x0040);
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		wm8711_write(codec, WM8711_ACTIVE, 0x0);
-+		wm8711_write(codec, WM8711_PWR, 0xffff);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8711_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-+		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-+		SNDRV_PCM_RATE_96000)
-+
-+#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8711_dai = {
-+	.name = "WM8711",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8711_RATES,
-+		.formats = WM8711_FORMATS,},
-+	.ops = {
-+		.prepare = wm8711_pcm_prepare,
-+		.hw_params = wm8711_hw_params,
-+		.shutdown = wm8711_shutdown,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8711_mute,
-+		.set_sysclk = wm8711_set_dai_sysclk,
-+		.set_fmt = wm8711_set_dai_fmt,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8711_dai);
-+
-+static int wm8711_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8711_write(codec, WM8711_ACTIVE, 0x0);
-+	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8711_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8711_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8711 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8711_init(struct snd_soc_device* socdev)
-+{
-+	struct snd_soc_codec* codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "WM8711";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8711_read_reg_cache;
-+	codec->write = wm8711_write;
-+	codec->dapm_event = wm8711_dapm_event;
-+	codec->dai = &wm8711_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8711_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8711_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8711_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8711_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8711_reg);
-+
-+	wm8711_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "wm8711: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* set the update bits */
-+	reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V);
-+	wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100);
-+	reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V);
-+	wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100);
-+
-+	wm8711_add_controls(codec);
-+	wm8711_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+    if (ret < 0) {
-+      	printk(KERN_ERR "wm8711: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8711_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8711 2 wire address is determined by GPIO5
-+ * state during powerup.
-+ *    low  = 0x1a
-+ *    high = 0x1b
-+ */
-+#define I2C_DRIVERID_WM8711 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8711_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8711_socdev;
-+	struct wm8711_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+
-+	i2c_set_clientdata(i2c, codec);
-+
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+        goto err;
-+    }
-+
-+	ret = wm8711_init(socdev);
-+    if (ret < 0) {
-+        err("failed to initialise WM8711\n");
-+        goto err;
-+    }
-+    return ret;
-+
-+err:
-+    kfree(codec);
-+    kfree(i2c);
-+    return ret;
-+
-+}
-+
-+static int wm8711_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8711_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8711_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8711_i2c_driver = {
-+	.driver = {
-+		.name = "WM8711 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8711,
-+	.attach_adapter = wm8711_i2c_attach,
-+	.detach_client =  wm8711_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8711",
-+	.driver = &wm8711_i2c_driver,
-+};
-+#endif
-+
-+static int wm8711_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8711_setup_data *setup;
-+	struct snd_soc_codec* codec;
-+	struct wm8711_priv *wm8711;
-+	int ret = 0;
-+
-+	info("WM8711 Audio Codec %s", WM8711_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
-+	if (wm8711 == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+
-+	codec->private_data = wm8711;
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8711_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8711_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8711_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8711_i2c_driver);
-+#endif
-+	kfree(codec->private_data);
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8711 = {
-+	.probe = 	wm8711_probe,
-+	.remove = 	wm8711_remove,
-+	.suspend = 	wm8711_suspend,
-+	.resume =	wm8711_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711);
-+
-+MODULE_DESCRIPTION("ASoC WM8711 driver");
-+MODULE_AUTHOR("Mike Arthur");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8711.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8711.h
-@@ -0,0 +1,42 @@
-+/*
-+ * wm8711.h  --  WM8711 Soc Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics
-+ *
-+ * Author: Mike Arthur <linux at wolfsonmicro.com>
-+ *
-+ * Based on wm8731.h
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8711_H
-+#define _WM8711_H
-+
-+/* WM8711 register space */
-+
-+#define WM8711_LOUT1V   0x02
-+#define WM8711_ROUT1V   0x03
-+#define WM8711_APANA    0x04
-+#define WM8711_APDIGI   0x05
-+#define WM8711_PWR      0x06
-+#define WM8711_IFACE    0x07
-+#define WM8711_SRATE    0x08
-+#define WM8711_ACTIVE   0x09
-+#define WM8711_RESET	0x0f
-+
-+#define WM8711_CACHEREGNUM 	8
-+
-+#define WM8711_SYSCLK	0
-+#define WM8711_DAI		0
-+
-+struct wm8711_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8711_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8711;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8980.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8980.c
-@@ -0,0 +1,923 @@
-+/*
-+ * wm8980.c  --  WM8980 ALSA Soc Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ *
-+ * Authors:
-+ * Mike Arthur      <linux at wolfsonmicro.com>
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8980.h"
-+
-+#define AUDIO_NAME "wm8980"
-+#define WM8980_VERSION "0.3"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8980_DEBUG 0
-+
-+#ifdef WM8980_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8980;
-+
-+/*
-+ * wm8980 register cache
-+ * We can't read the WM8980 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8980_reg[WM8980_CACHEREGNUM] = {
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0050, 0x0000, 0x0140, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x00ff,
-+    0x00ff, 0x0000, 0x0100, 0x00ff,
-+    0x00ff, 0x0000, 0x012c, 0x002c,
-+    0x002c, 0x002c, 0x002c, 0x0000,
-+    0x0032, 0x0000, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0038, 0x000b, 0x0032, 0x0000,
-+    0x0008, 0x000c, 0x0093, 0x00e9,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0033, 0x0010, 0x0010, 0x0100,
-+    0x0100, 0x0002, 0x0001, 0x0001,
-+    0x0039, 0x0039, 0x0039, 0x0039,
-+    0x0001, 0x0001,
-+};
-+
-+/*
-+ * read wm8980 register cache
-+ */
-+static inline unsigned int wm8980_read_reg_cache(struct snd_soc_codec  *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8980_RESET)
-+		return 0;
-+	if (reg >= WM8980_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8980 register cache
-+ */
-+static inline void wm8980_write_reg_cache(struct snd_soc_codec  *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8980_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8980 register space
-+ */
-+static int wm8980_write(struct snd_soc_codec  *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8980 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8980_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -1;
-+}
-+
-+#define wm8980_reset(c)	wm8980_write(c, WM8980_RESET, 0)
-+
-+static const char *wm8980_companding[] = {"Off", "NC", "u-law", "A-law" };
-+static const char *wm8980_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
-+static const char *wm8980_eqmode[] = {"Capture", "Playback" };
-+static const char *wm8980_bw[] = {"Narrow", "Wide" };
-+static const char *wm8980_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
-+static const char *wm8980_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
-+static const char *wm8980_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
-+static const char *wm8980_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
-+static const char *wm8980_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
-+static const char *wm8980_alc[] =
-+    {"ALC both on", "ALC left only", "ALC right only", "Limiter" };
-+
-+static const struct soc_enum wm8980_enum[] = {
-+	SOC_ENUM_SINGLE(WM8980_COMP, 1, 4, wm8980_companding), /* adc */
-+	SOC_ENUM_SINGLE(WM8980_COMP, 3, 4, wm8980_companding), /* dac */
-+	SOC_ENUM_SINGLE(WM8980_DAC,  4, 4, wm8980_deemp),
-+	SOC_ENUM_SINGLE(WM8980_EQ1,  8, 2, wm8980_eqmode),
-+
-+	SOC_ENUM_SINGLE(WM8980_EQ1,  5, 4, wm8980_eq1),
-+	SOC_ENUM_SINGLE(WM8980_EQ2,  8, 2, wm8980_bw),
-+	SOC_ENUM_SINGLE(WM8980_EQ2,  5, 4, wm8980_eq2),
-+	SOC_ENUM_SINGLE(WM8980_EQ3,  8, 2, wm8980_bw),
-+
-+	SOC_ENUM_SINGLE(WM8980_EQ3,  5, 4, wm8980_eq3),
-+	SOC_ENUM_SINGLE(WM8980_EQ4,  8, 2, wm8980_bw),
-+	SOC_ENUM_SINGLE(WM8980_EQ4,  5, 4, wm8980_eq4),
-+	SOC_ENUM_SINGLE(WM8980_EQ5,  8, 2, wm8980_bw),
-+
-+	SOC_ENUM_SINGLE(WM8980_EQ5,  5, 4, wm8980_eq5),
-+	SOC_ENUM_SINGLE(WM8980_ALC3,  8, 2, wm8980_alc),
-+};
-+
-+static const struct snd_kcontrol_new wm8980_snd_controls[] = {
-+SOC_SINGLE("Digital Loopback Switch", WM8980_COMP, 0, 1, 0),
-+
-+SOC_ENUM("ADC Companding", wm8980_enum[0]),
-+SOC_ENUM("DAC Companding", wm8980_enum[1]),
-+
-+SOC_SINGLE("Jack Detection Enable", WM8980_JACK1, 6, 1, 0),
-+
-+SOC_SINGLE("DAC Right Inversion Switch", WM8980_DAC, 1, 1, 0),
-+SOC_SINGLE("DAC Left Inversion Switch", WM8980_DAC, 0, 1, 0),
-+
-+SOC_SINGLE("Left Playback Volume", WM8980_DACVOLL, 0, 127, 0),
-+SOC_SINGLE("Right Playback Volume", WM8980_DACVOLR, 0, 127, 0),
-+
-+SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Cut Off", WM8980_ADC, 4, 7, 0),
-+SOC_SINGLE("Right ADC Inversion Switch", WM8980_ADC, 1, 1, 0),
-+SOC_SINGLE("Left ADC Inversion Switch", WM8980_ADC, 0, 1, 0),
-+
-+SOC_SINGLE("Left Capture Volume", WM8980_ADCVOLL,  0, 127, 0),
-+SOC_SINGLE("Right Capture Volume", WM8980_ADCVOLR,  0, 127, 0),
-+
-+SOC_ENUM("Equaliser Function", wm8980_enum[3]),
-+SOC_ENUM("EQ1 Cut Off", wm8980_enum[4]),
-+SOC_SINGLE("EQ1 Volume", WM8980_EQ1,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ2 Bandwith", wm8980_enum[5]),
-+SOC_ENUM("EQ2 Cut Off", wm8980_enum[6]),
-+SOC_SINGLE("EQ2 Volume", WM8980_EQ2,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ3 Bandwith", wm8980_enum[7]),
-+SOC_ENUM("EQ3 Cut Off", wm8980_enum[8]),
-+SOC_SINGLE("EQ3 Volume", WM8980_EQ3,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ4 Bandwith", wm8980_enum[9]),
-+SOC_ENUM("EQ4 Cut Off", wm8980_enum[10]),
-+SOC_SINGLE("EQ4 Volume", WM8980_EQ4,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ5 Bandwith", wm8980_enum[11]),
-+SOC_ENUM("EQ5 Cut Off", wm8980_enum[12]),
-+SOC_SINGLE("EQ5 Volume", WM8980_EQ5,  0, 31, 1),
-+
-+SOC_SINGLE("DAC Playback Limiter Switch", WM8980_DACLIM1,  8, 1, 0),
-+SOC_SINGLE("DAC Playback Limiter Decay", WM8980_DACLIM1,  4, 15, 0),
-+SOC_SINGLE("DAC Playback Limiter Attack", WM8980_DACLIM1,  0, 15, 0),
-+
-+SOC_SINGLE("DAC Playback Limiter Threshold", WM8980_DACLIM2,  4, 7, 0),
-+SOC_SINGLE("DAC Playback Limiter Boost", WM8980_DACLIM2,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Enable Switch", WM8980_ALC1,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Max Gain", WM8980_ALC1,  3, 7, 0),
-+SOC_SINGLE("ALC Capture Min Gain", WM8980_ALC1,  0, 7, 0),
-+
-+SOC_SINGLE("ALC Capture ZC Switch", WM8980_ALC2,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Hold", WM8980_ALC2,  4, 7, 0),
-+SOC_SINGLE("ALC Capture Target", WM8980_ALC2,  0, 15, 0),
-+
-+SOC_ENUM("ALC Capture Mode", wm8980_enum[13]),
-+SOC_SINGLE("ALC Capture Decay", WM8980_ALC3,  4, 15, 0),
-+SOC_SINGLE("ALC Capture Attack", WM8980_ALC3,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8980_NGATE,  3, 1, 0),
-+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8980_NGATE,  0, 7, 0),
-+
-+SOC_SINGLE("Left Capture PGA ZC Switch", WM8980_INPPGAL,  7, 1, 0),
-+SOC_SINGLE("Left Capture PGA Volume", WM8980_INPPGAL,  0, 63, 0),
-+
-+SOC_SINGLE("Right Capture PGA ZC Switch", WM8980_INPPGAR,  7, 1, 0),
-+SOC_SINGLE("Right Capture PGA Volume", WM8980_INPPGAR,  0, 63, 0),
-+
-+SOC_SINGLE("Left Headphone Playback ZC Switch", WM8980_HPVOLL,  7, 1, 0),
-+SOC_SINGLE("Left Headphone Playback Switch", WM8980_HPVOLL,  6, 1, 1),
-+SOC_SINGLE("Left Headphone Playback Volume", WM8980_HPVOLL,  0, 63, 0),
-+
-+SOC_SINGLE("Right Headphone Playback ZC Switch", WM8980_HPVOLR,  7, 1, 0),
-+SOC_SINGLE("Right Headphone Playback Switch", WM8980_HPVOLR,  6, 1, 1),
-+SOC_SINGLE("Right Headphone Playback Volume", WM8980_HPVOLR,  0, 63, 0),
-+
-+SOC_SINGLE("Left Speaker Playback ZC Switch", WM8980_SPKVOLL,  7, 1, 0),
-+SOC_SINGLE("Left Speaker Playback Switch", WM8980_SPKVOLL,  6, 1, 1),
-+SOC_SINGLE("Left Speaker Playback Volume", WM8980_SPKVOLL,  0, 63, 0),
-+
-+SOC_SINGLE("Right Speaker Playback ZC Switch", WM8980_SPKVOLR,  7, 1, 0),
-+SOC_SINGLE("Right Speaker Playback Switch", WM8980_SPKVOLR,  6, 1, 1),
-+SOC_SINGLE("Right Speaker Playback Volume", WM8980_SPKVOLR,  0, 63, 0),
-+
-+SOC_DOUBLE_R("Capture Boost(+20dB)", WM8980_ADCBOOSTL, WM8980_ADCBOOSTR,
-+	8, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8980_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8980_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8980_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Left Output Mixer */
-+static const snd_kcontrol_new_t wm8980_left_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_OUTPUT, 6, 1, 1),
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_MIXL, 0, 1, 1),
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXL, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXL, 5, 1, 0),
-+};
-+
-+/* Right Output Mixer */
-+static const snd_kcontrol_new_t wm8980_right_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_OUTPUT, 5, 1, 1),
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_MIXR, 0, 1, 1),
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXR, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXR, 5, 1, 0),
-+};
-+
-+/* Left AUX Input boost vol */
-+static const snd_kcontrol_new_t wm8980_laux_boost_controls =
-+SOC_DAPM_SINGLE("Left Aux Volume", WM8980_ADCBOOSTL, 0, 3, 0);
-+
-+/* Right AUX Input boost vol */
-+static const snd_kcontrol_new_t wm8980_raux_boost_controls =
-+SOC_DAPM_SINGLE("Right Aux Volume", WM8980_ADCBOOSTR, 0, 3, 0);
-+
-+/* Left Input boost vol */
-+static const snd_kcontrol_new_t wm8980_lmic_boost_controls =
-+SOC_DAPM_SINGLE("Left Input Volume", WM8980_ADCBOOSTL, 4, 3, 0);
-+
-+/* Right Input boost vol */
-+static const snd_kcontrol_new_t wm8980_rmic_boost_controls =
-+SOC_DAPM_SINGLE("Right Input Volume", WM8980_ADCBOOSTR, 4, 3, 0);
-+
-+/* Left Aux In to PGA */
-+static const snd_kcontrol_new_t wm8980_laux_capture_boost_controls =
-+SOC_DAPM_SINGLE("Left Capture Switch", WM8980_ADCBOOSTL,  8, 1, 0);
-+
-+/* Right  Aux In to PGA */
-+static const snd_kcontrol_new_t wm8980_raux_capture_boost_controls =
-+SOC_DAPM_SINGLE("Right Capture Switch", WM8980_ADCBOOSTR,  8, 1, 0);
-+
-+/* Left Input P In to PGA */
-+static const snd_kcontrol_new_t wm8980_lmicp_capture_boost_controls =
-+SOC_DAPM_SINGLE("Left Input P Capture Boost Switch", WM8980_INPUT,  0, 1, 0);
-+
-+/* Right Input P In to PGA */
-+static const snd_kcontrol_new_t wm8980_rmicp_capture_boost_controls =
-+SOC_DAPM_SINGLE("Right Input P Capture Boost Switch", WM8980_INPUT,  4, 1, 0);
-+
-+/* Left Input N In to PGA */
-+static const snd_kcontrol_new_t wm8980_lmicn_capture_boost_controls =
-+SOC_DAPM_SINGLE("Left Input N Capture Boost Switch", WM8980_INPUT,  1, 1, 0);
-+
-+/* Right Input N In to PGA */
-+static const snd_kcontrol_new_t wm8980_rmicn_capture_boost_controls =
-+SOC_DAPM_SINGLE("Right Input N Capture Boost Switch", WM8980_INPUT,  5, 1, 0);
-+
-+// TODO Widgets
-+static const struct snd_soc_dapm_widget wm8980_dapm_widgets[] = {
-+#if 0
-+//SND_SOC_DAPM_MUTE("Mono Mute", WM8980_MONOMIX, 6, 0),
-+//SND_SOC_DAPM_MUTE("Speaker Mute", WM8980_SPKMIX, 6, 0),
-+
-+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8980_POWER3, 2, 0,
-+	&wm8980_speaker_mixer_controls[0],
-+	ARRAY_SIZE(wm8980_speaker_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Mono Mixer", WM8980_POWER3, 3, 0,
-+	&wm8980_mono_mixer_controls[0],
-+	ARRAY_SIZE(wm8980_mono_mixer_controls)),
-+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8980_POWER3, 0, 0),
-+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8980_POWER3, 0, 0),
-+SND_SOC_DAPM_PGA("Aux Input", WM8980_POWER1, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkN Out", WM8980_POWER3, 5, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkP Out", WM8980_POWER3, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out", WM8980_POWER3, 7, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic PGA", WM8980_POWER2, 2, 0, NULL, 0),
-+
-+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8980_aux_boost_controls, 1),
-+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8980_mic_boost_controls, 1),
-+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8980_capture_boost_controls),
-+
-+SND_SOC_DAPM_MIXER("Boost Mixer", WM8980_POWER2, 4, 0, NULL, 0),
-+
-+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8980_POWER1, 4, 0),
-+
-+SND_SOC_DAPM_INPUT("MICN"),
-+SND_SOC_DAPM_INPUT("MICP"),
-+SND_SOC_DAPM_INPUT("AUX"),
-+SND_SOC_DAPM_OUTPUT("MONOOUT"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
-+#endif
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* Mono output mixer */
-+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
-+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Speaker output mixer */
-+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
-+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Outputs */
-+	{"Mono Out", NULL, "Mono Mixer"},
-+	{"MONOOUT", NULL, "Mono Out"},
-+	{"SpkN Out", NULL, "Speaker Mixer"},
-+	{"SpkP Out", NULL, "Speaker Mixer"},
-+	{"SPKOUTN", NULL, "SpkN Out"},
-+	{"SPKOUTP", NULL, "SpkP Out"},
-+
-+	/* Boost Mixer */
-+	{"Boost Mixer", NULL, "ADC"},
-+	{"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"},
-+	{"Aux Boost", "Aux Volume", "Boost Mixer"},
-+	{"Capture Boost", "Capture Switch", "Boost Mixer"},
-+	{"Mic Boost", "Mic Volume", "Boost Mixer"},
-+
-+	/* Inputs */
-+	{"MICP", NULL, "Mic Boost"},
-+	{"MICN", NULL, "Mic PGA"},
-+	{"Mic PGA", NULL, "Capture Boost"},
-+	{"AUX", NULL, "Aux Input"},
-+
-+    /*  */
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8980_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8980_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8980_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path map */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1],
-+            audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct pll_ {
-+	unsigned int in_hz, out_hz;
-+	unsigned int pre:4; /* prescale - 1 */
-+	unsigned int n:4;
-+	unsigned int k;
-+};
-+
-+struct pll_ pll[] = {
-+	{12000000, 11289600, 0, 7, 0x86c220},
-+	{12000000, 12288000, 0, 8, 0x3126e8},
-+	{13000000, 11289600, 0, 6, 0xf28bd4},
-+	{13000000, 12288000, 0, 7, 0x8fd525},
-+	{12288000, 11289600, 0, 7, 0x59999a},
-+	{11289600, 12288000, 0, 8, 0x80dee9},
-+	/* TODO: liam - add more entries */
-+};
-+
-+static int wm8980_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	int i;
-+	u16 reg;
-+
-+	if(freq_in == 0 || freq_out == 0) {
-+		reg = wm8980_read_reg_cache(codec, WM8980_POWER1);
-+		wm8980_write(codec, WM8980_POWER1, reg & 0x1df);
-+		return 0;
-+	}
-+
-+	for(i = 0; i < ARRAY_SIZE(pll); i++) {
-+		if (freq_in == pll[i].in_hz && freq_out == pll[i].out_hz) {
-+			wm8980_write(codec, WM8980_PLLN, (pll[i].pre << 4) | pll[i].n);
-+			wm8980_write(codec, WM8980_PLLK1, pll[i].k >> 18);
-+			wm8980_write(codec, WM8980_PLLK1, (pll[i].k >> 9) && 0x1ff);
-+			wm8980_write(codec, WM8980_PLLK1, pll[i].k && 0x1ff);
-+			reg = wm8980_read_reg_cache(codec, WM8980_POWER1);
-+			wm8980_write(codec, WM8980_POWER1, reg | 0x020);
-+			return 0;
-+		}
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8980_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = wm8980_read_reg_cache(codec, WM8980_IFACE) & 0x3;
-+	u16 clk = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0xfffe;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		clk |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0008;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x00018;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0180;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0100;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0080;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8980_write(codec, WM8980_IFACE, iface);
-+	return 0;
-+}
-+
-+static int wm8980_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 iface = wm8980_read_reg_cache(codec, WM8980_IFACE) & 0xff9f;
-+	u16 adn = wm8980_read_reg_cache(codec, WM8980_ADD) & 0x1f1;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0020;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0040;
-+		break;
-+	}
-+
-+	/* filter coefficient */
-+	switch (params_rate(params)) {
-+	case SNDRV_PCM_RATE_8000:
-+		adn |= 0x5 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_11025:
-+		adn |= 0x4 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_16000:
-+		adn |= 0x3 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_22050:
-+		adn |= 0x2 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_32000:
-+		adn |= 0x1 << 1;
-+		break;
-+	}
-+
-+	/* set iface */
-+	wm8980_write(codec, WM8980_IFACE, iface);
-+	wm8980_write(codec, WM8980_ADD, adn);
-+	return 0;
-+}
-+
-+static int wm8980_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8980_MCLKDIV:
-+		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x11f;
-+		wm8980_write(codec, WM8980_CLOCK, reg | div);
-+		break;
-+	case WM8980_BCLKDIV:
-+		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x1c7;
-+		wm8980_write(codec, WM8980_CLOCK, reg | div);
-+		break;
-+	case WM8980_OPCLKDIV:
-+		reg = wm8980_read_reg_cache(codec, WM8980_GPIO) & 0x1cf;
-+		wm8980_write(codec, WM8980_GPIO, reg | div);
-+		break;
-+	case WM8980_DACOSR:
-+		reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0x1f7;
-+		wm8980_write(codec, WM8980_DAC, reg | div);
-+		break;
-+	case WM8980_ADCOSR:
-+		reg = wm8980_read_reg_cache(codec, WM8980_ADC) & 0x1f7;
-+		wm8980_write(codec, WM8980_ADC, reg | div);
-+		break;
-+	case WM8980_MCLKSEL:
-+		reg = wm8980_read_reg_cache(codec, WM8980_CLOCK) & 0x0ff;
-+		wm8980_write(codec, WM8980_CLOCK, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8980_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0xffbf;
-+
-+	if(mute)
-+		wm8980_write(codec, WM8980_DAC, mute_reg | 0x40);
-+	else
-+		wm8980_write(codec, WM8980_DAC, mute_reg);
-+
-+	return 0;
-+}
-+
-+/* TODO: liam need to make this lower power with dapm */
-+static int wm8980_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, clk and osc on, dac unmute, active */
-+		wm8980_write(codec, WM8980_POWER1, 0x1ff);
-+		wm8980_write(codec, WM8980_POWER2, 0x1ff);
-+		wm8980_write(codec, WM8980_POWER3, 0x1ff);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, dac mute, inactive */
-+
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		wm8980_write(codec, WM8980_POWER1, 0x0);
-+		wm8980_write(codec, WM8980_POWER2, 0x0);
-+		wm8980_write(codec, WM8980_POWER3, 0x0);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8980_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000)
-+
-+#define WM8980_FORMATS \
-+	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
-+	SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8980_dai = {
-+	.name = "WM8980 HiFi",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8980_RATES,
-+		.formats = WM8980_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8980_RATES,
-+		.formats = WM8980_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8980_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8980_mute,
-+		.set_fmt = wm8980_set_dai_fmt,
-+		.set_clkdiv = wm8980_set_dai_clkdiv,
-+		.set_pll = wm8980_set_dai_pll,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8980_dai);
-+
-+static int wm8980_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8980_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8980_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8980_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8980 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8980_init(struct snd_soc_device* socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "WM8980";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8980_read_reg_cache;
-+	codec->write = wm8980_write;
-+	codec->dapm_event = wm8980_dapm_event;
-+	codec->dai = &wm8980_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8980_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8980_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8980_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8980_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8980_reg);
-+
-+	wm8980_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "wm8980: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8980_add_controls(codec);
-+	wm8980_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8980: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8980_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8980 2 wire address is 0x1a
-+ */
-+#define I2C_DRIVERID_WM8980 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8980_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8980_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8980_socdev;
-+	struct wm8980_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+
-+	i2c_set_clientdata(i2c, codec);
-+
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if(ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8980_init(socdev);
-+	if(ret < 0) {
-+		err("failed to initialise WM8980\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+
-+}
-+
-+static int wm8980_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8980_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8980_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8980_i2c_driver = {
-+	.driver = {
-+		.name = "WM8980 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8980,
-+	.attach_adapter = wm8980_i2c_attach,
-+	.detach_client =  wm8980_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8980",
-+	.driver = &wm8980_i2c_driver,
-+};
-+#endif
-+
-+static int wm8980_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8980_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("WM8980 Audio Codec %s", WM8980_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8980_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8980_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8980_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8980_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8980 = {
-+	.probe = 	wm8980_probe,
-+	.remove = 	wm8980_remove,
-+	.suspend = 	wm8980_suspend,
-+	.resume =	wm8980_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8980);
-+
-+MODULE_DESCRIPTION("ASoC WM8980 driver");
-+MODULE_AUTHOR("Mike Arthur");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8980.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8980.h
-@@ -0,0 +1,116 @@
-+/*
-+ * wm8980.h  --  WM8980 Soc Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8980_H
-+#define _WM8980_H
-+
-+/* WM8980 register space */
-+
-+#define WM8980_RESET		0x0
-+#define WM8980_POWER1		0x1
-+#define WM8980_POWER2		0x2
-+#define WM8980_POWER3		0x3
-+#define WM8980_IFACE		0x4
-+#define WM8980_COMP			0x5
-+#define WM8980_CLOCK		0x6
-+#define WM8980_ADD			0x7
-+#define WM8980_GPIO			0x8
-+#define WM8980_JACK1        0x9
-+#define WM8980_DAC			0xa
-+#define WM8980_DACVOLL	    0xb
-+#define WM8980_DACVOLR      0xc
-+#define WM8980_JACK2        0xd
-+#define WM8980_ADC			0xe
-+#define WM8980_ADCVOLL		0xf
-+#define WM8980_ADCVOLR      0x10
-+#define WM8980_EQ1			0x12
-+#define WM8980_EQ2			0x13
-+#define WM8980_EQ3			0x14
-+#define WM8980_EQ4			0x15
-+#define WM8980_EQ5			0x16
-+#define WM8980_DACLIM1		0x18
-+#define WM8980_DACLIM2		0x19
-+#define WM8980_NOTCH1		0x1b
-+#define WM8980_NOTCH2		0x1c
-+#define WM8980_NOTCH3		0x1d
-+#define WM8980_NOTCH4		0x1e
-+#define WM8980_ALC1			0x20
-+#define WM8980_ALC2			0x21
-+#define WM8980_ALC3			0x22
-+#define WM8980_NGATE		0x23
-+#define WM8980_PLLN			0x24
-+#define WM8980_PLLK1		0x25
-+#define WM8980_PLLK2		0x26
-+#define WM8980_PLLK3		0x27
-+#define WM8980_VIDEO		0x28
-+#define WM8980_3D           0x29
-+#define WM8980_BEEP         0x2b
-+#define WM8980_INPUT		0x2c
-+#define WM8980_INPPGAL  	0x2d
-+#define WM8980_INPPGAR      0x2e
-+#define WM8980_ADCBOOSTL	0x2f
-+#define WM8980_ADCBOOSTR    0x30
-+#define WM8980_OUTPUT		0x31
-+#define WM8980_MIXL	        0x32
-+#define WM8980_MIXR         0x33
-+#define WM8980_HPVOLL		0x34
-+#define WM8980_HPVOLR       0x35
-+#define WM8980_SPKVOLL      0x36
-+#define WM8980_SPKVOLR      0x37
-+#define WM8980_OUT3MIX		0x38
-+#define WM8980_MONOMIX      0x39
-+
-+#define WM8980_CACHEREGNUM 	58
-+
-+/*
-+ * WM8980 Clock dividers
-+ */
-+#define WM8980_MCLKDIV 		0
-+#define WM8980_BCLKDIV		1
-+#define WM8980_OPCLKDIV		2
-+#define WM8980_DACOSR		3
-+#define WM8980_ADCOSR		4
-+#define WM8980_MCLKSEL		5
-+
-+#define WM8980_MCLK_MCLK		(0 << 8)
-+#define WM8980_MCLK_PLL			(1 << 8)
-+
-+#define WM8980_MCLK_DIV_1		(0 << 5)
-+#define WM8980_MCLK_DIV_1_5		(1 << 5)
-+#define WM8980_MCLK_DIV_2		(2 << 5)
-+#define WM8980_MCLK_DIV_3		(3 << 5)
-+#define WM8980_MCLK_DIV_4		(4 << 5)
-+#define WM8980_MCLK_DIV_5_5		(5 << 5)
-+#define WM8980_MCLK_DIV_6		(6 << 5)
-+
-+#define WM8980_BCLK_DIV_1		(0 << 2)
-+#define WM8980_BCLK_DIV_2		(1 << 2)
-+#define WM8980_BCLK_DIV_4		(2 << 2)
-+#define WM8980_BCLK_DIV_8		(3 << 2)
-+#define WM8980_BCLK_DIV_16		(4 << 2)
-+#define WM8980_BCLK_DIV_32		(5 << 2)
-+
-+#define WM8980_DACOSR_64		(0 << 3)
-+#define WM8980_DACOSR_128		(1 << 3)
-+
-+#define WM8980_ADCOSR_64		(0 << 3)
-+#define WM8980_ADCOSR_128		(1 << 3)
-+
-+#define WM8980_OPCLK_DIV_1		(0 << 4)
-+#define WM8980_OPCLK_DIV_2		(1 << 4)
-+#define WM8980_OPCLK_DIV_3		(2 << 4)
-+#define WM8980_OPCLK_DIV_4		(3 << 4)
-+
-+struct wm8980_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8980_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8980;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8510.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8510.c
-@@ -0,0 +1,860 @@
-+/*
-+ * wm8510.c  --  WM8510 ALSA Soc Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ *
-+ * Author: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8510.h"
-+
-+#define AUDIO_NAME "wm8510"
-+#define WM8510_VERSION "0.6"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8510_DEBUG 0
-+
-+#ifdef WM8510_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8510;
-+
-+/*
-+ * wm8510 register cache
-+ * We can't read the WM8510 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0050, 0x0000, 0x0140, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x00ff,
-+    0x0000, 0x0000, 0x0100, 0x00ff,
-+    0x0000, 0x0000, 0x012c, 0x002c,
-+    0x002c, 0x002c, 0x002c, 0x0000,
-+    0x0032, 0x0000, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0038, 0x000b, 0x0032, 0x0000,
-+    0x0008, 0x000c, 0x0093, 0x00e9,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0003, 0x0010, 0x0000, 0x0000,
-+    0x0000, 0x0002, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0039, 0x0000,
-+    0x0000,
-+};
-+
-+/*
-+ * read wm8510 register cache
-+ */
-+static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec * codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8510_RESET)
-+		return 0;
-+	if (reg >= WM8510_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8510 register cache
-+ */
-+static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8510_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8510 register space
-+ */
-+static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8510 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8510_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8510_reset(c)	wm8510_write(c, WM8510_RESET, 0)
-+
-+static const char *wm8510_companding[] = {"Off", "NC", "u-law", "A-law" };
-+static const char *wm8510_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
-+static const char *wm8510_alc[] = {"ALC", "Limiter" };
-+
-+static const struct soc_enum wm8510_enum[] = {
-+	SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */
-+	SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */
-+	SOC_ENUM_SINGLE(WM8510_DAC,  4, 4, wm8510_deemp),
-+	SOC_ENUM_SINGLE(WM8510_ALC3,  8, 2, wm8510_alc),
-+};
-+
-+static const struct snd_kcontrol_new wm8510_snd_controls[] = {
-+
-+SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0),
-+
-+SOC_ENUM("DAC Companding", wm8510_enum[1]),
-+SOC_ENUM("ADC Companding", wm8510_enum[0]),
-+
-+SOC_ENUM("Playback De-emphasis", wm8510_enum[2]),
-+SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0),
-+
-+SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0),
-+
-+SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0),
-+SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0),
-+
-+SOC_SINGLE("Capture Volume", WM8510_ADCVOL,  0, 127, 0),
-+
-+SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1,  8, 1, 0),
-+SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1,  4, 15, 0),
-+SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1,  0, 15, 0),
-+
-+SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2,  4, 7, 0),
-+SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Enable Switch", WM8510_ALC1,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1,  3, 7, 0),
-+SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1,  0, 7, 0),
-+
-+SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Hold", WM8510_ALC2,  4, 7, 0),
-+SOC_SINGLE("ALC Capture Target", WM8510_ALC2,  0, 15, 0),
-+
-+SOC_ENUM("ALC Capture Mode", wm8510_enum[3]),
-+SOC_SINGLE("ALC Capture Decay", WM8510_ALC3,  4, 15, 0),
-+SOC_SINGLE("ALC Capture Attack", WM8510_ALC3,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE,  3, 1, 0),
-+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE,  0, 7, 0),
-+
-+SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA,  7, 1, 0),
-+SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA,  0, 63, 0),
-+
-+SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL,  7, 1, 0),
-+SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL,  6, 1, 1),
-+SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL,  0, 63, 0),
-+
-+SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
-+SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8510_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8510_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Speaker Output Mixer */
-+static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1),
-+};
-+
-+/* Mono Output Mixer */
-+static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0),
-+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1),
-+};
-+
-+/* AUX Input boost vol */
-+static const struct snd_kcontrol_new wm8510_aux_boost_controls =
-+SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0);
-+
-+/* Mic Input boost vol */
-+static const struct snd_kcontrol_new wm8510_mic_boost_controls =
-+SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0);
-+
-+/* Capture boost switch */
-+static const struct snd_kcontrol_new wm8510_capture_boost_controls =
-+SOC_DAPM_SINGLE("Capture Boost Switch", WM8510_INPPGA,  6, 1, 0);
-+
-+/* Aux In to PGA */
-+static const struct snd_kcontrol_new wm8510_aux_capture_boost_controls =
-+SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8510_INPPGA,  2, 1, 0);
-+
-+/* Mic P In to PGA */
-+static const struct snd_kcontrol_new wm8510_micp_capture_boost_controls =
-+SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8510_INPPGA,  0, 1, 0);
-+
-+/* Mic N In to PGA */
-+static const struct snd_kcontrol_new wm8510_micn_capture_boost_controls =
-+SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8510_INPPGA,  1, 1, 0);
-+
-+static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = {
-+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0,
-+	&wm8510_speaker_mixer_controls[0],
-+	ARRAY_SIZE(wm8510_speaker_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0,
-+	&wm8510_mono_mixer_controls[0],
-+	ARRAY_SIZE(wm8510_mono_mixer_controls)),
-+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0),
-+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0),
-+SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, NULL, 0),
-+
-+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8510_aux_boost_controls, 1),
-+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8510_mic_boost_controls, 1),
-+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8510_capture_boost_controls),
-+
-+SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, NULL, 0),
-+
-+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0),
-+
-+SND_SOC_DAPM_INPUT("MICN"),
-+SND_SOC_DAPM_INPUT("MICP"),
-+SND_SOC_DAPM_INPUT("AUX"),
-+SND_SOC_DAPM_OUTPUT("MONOOUT"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* Mono output mixer */
-+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
-+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Speaker output mixer */
-+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
-+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Outputs */
-+	{"Mono Out", NULL, "Mono Mixer"},
-+	{"MONOOUT", NULL, "Mono Out"},
-+	{"SpkN Out", NULL, "Speaker Mixer"},
-+	{"SpkP Out", NULL, "Speaker Mixer"},
-+	{"SPKOUTN", NULL, "SpkN Out"},
-+	{"SPKOUTP", NULL, "SpkP Out"},
-+
-+	/* Boost Mixer */
-+	{"Boost Mixer", NULL, "ADC"},
-+    {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"},
-+	{"Aux Boost", "Aux Volume", "Boost Mixer"},
-+    {"Capture Boost", "Capture Switch", "Boost Mixer"},
-+	{"Mic Boost", "Mic Volume", "Boost Mixer"},
-+
-+	/* Inputs */
-+	{"MICP", NULL, "Mic Boost"},
-+	{"MICN", NULL, "Mic PGA"},
-+	{"Mic PGA", NULL, "Capture Boost"},
-+	{"AUX", NULL, "Aux Input"},
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8510_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8510_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8510_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct pll_ {
-+	unsigned int pre_div:4; /* prescale - 1 */
-+	unsigned int n:4;
-+	unsigned int k;
-+};
-+
-+static struct pll_ pll_div;
-+
-+/* The size in bits of the pll divide multiplied by 10
-+ * to allow rounding later */
-+#define FIXED_PLL_SIZE ((1 << 24) * 10)
-+
-+static void pll_factors(unsigned int target, unsigned int source)
-+{
-+	unsigned long long Kpart;
-+	unsigned int K, Ndiv, Nmod;
-+
-+	Ndiv = target / source;
-+	if (Ndiv < 6) {
-+		source >>= 1;
-+		pll_div.pre_div = 1;
-+		Ndiv = target / source;
-+	} else
-+		pll_div.pre_div = 0;
-+
-+	if ((Ndiv < 6) || (Ndiv > 12))
-+		printk(KERN_WARNING
-+			"WM8510 N value outwith recommended range! N = %d\n",Ndiv);
-+
-+	pll_div.n = Ndiv;
-+	Nmod = target % source;
-+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
-+
-+	do_div(Kpart, source);
-+
-+	K = Kpart & 0xFFFFFFFF;
-+
-+	/* Check if we need to round */
-+	if ((K % 10) >= 5)
-+		K += 5;
-+
-+	/* Move down to proper range now rounding is done */
-+	K /= 10;
-+
-+	pll_div.k = K;
-+}
-+
-+static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	if(freq_in == 0 || freq_out == 0) {
-+		reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
-+		wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
-+		return 0;
-+	}
-+
-+	pll_factors(freq_out*8, freq_in);
-+
-+	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
-+	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
-+	wm8510_write(codec, WM8510_PLLK1, (pll_div.k >> 9) && 0x1ff);
-+	wm8510_write(codec, WM8510_PLLK1, pll_div.k && 0x1ff);
-+	reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
-+	wm8510_write(codec, WM8510_POWER1, reg | 0x020);
-+	return 0;
-+
-+}
-+
-+/*
-+ * Configure WM8510 clock dividers.
-+ */
-+static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8510_OPCLKDIV:
-+		reg = wm8510_read_reg_cache(codec, WM8510_GPIO & 0x1cf);
-+		wm8510_write(codec, WM8510_GPIO, reg | div);
-+		break;
-+	case WM8510_MCLKDIV:
-+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1f);
-+		wm8510_write(codec, WM8510_CLOCK, reg | div);
-+		break;
-+	case WM8510_ADCCLK:
-+		reg = wm8510_read_reg_cache(codec, WM8510_ADC & 0x1f7);
-+		wm8510_write(codec, WM8510_ADC, reg | div);
-+		break;
-+	case WM8510_DACCLK:
-+		reg = wm8510_read_reg_cache(codec, WM8510_DAC & 0x1f7);
-+		wm8510_write(codec, WM8510_DAC, reg | div);
-+		break;
-+	case WM8510_BCLKDIV:
-+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1e3);
-+		wm8510_write(codec, WM8510_CLOCK, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = 0;
-+	u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		clk |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0008;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x00018;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0180;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0100;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0080;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8510_write(codec, WM8510_IFACE, iface);
-+	wm8510_write(codec, WM8510_CLOCK, clk);
-+	return 0;
-+}
-+
-+static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 iface = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x19f;
-+	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0020;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0040;
-+		break;
-+	case SNDRV_PCM_FORMAT_S32_LE:
-+		iface |= 0x0060;
-+		break;
-+	}
-+
-+	/* filter coefficient */
-+	switch (params_rate(params)) {
-+	case SNDRV_PCM_RATE_8000:
-+		adn |= 0x5 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_11025:
-+		adn |= 0x4 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_16000:
-+		adn |= 0x3 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_22050:
-+		adn |= 0x2 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_32000:
-+		adn |= 0x1 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_44100:
-+		break;
-+	}
-+
-+	wm8510_write(codec, WM8510_IFACE, iface);
-+	wm8510_write(codec, WM8510_ADD, adn);
-+	return 0;
-+}
-+
-+static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
-+
-+	if(mute)
-+		wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
-+	else
-+		wm8510_write(codec, WM8510_DAC, mute_reg);
-+	return 0;
-+}
-+
-+/* liam need to make this lower power with dapm */
-+static int wm8510_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, clk and osc on, dac unmute, active */
-+		wm8510_write(codec, WM8510_POWER1, 0x1ff);
-+		wm8510_write(codec, WM8510_POWER2, 0x1ff);
-+		wm8510_write(codec, WM8510_POWER3, 0x1ff);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, dac mute, inactive */
-+
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		wm8510_write(codec, WM8510_POWER1, 0x0);
-+		wm8510_write(codec, WM8510_POWER2, 0x0);
-+		wm8510_write(codec, WM8510_POWER3, 0x0);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+		SNDRV_PCM_RATE_48000)
-+
-+#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8510_dai = {
-+	.name = "WM8510 HiFi",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8510_RATES,
-+		.formats = WM8510_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8510_RATES,
-+		.formats = WM8510_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8510_pcm_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8510_mute,
-+		.set_fmt = wm8510_set_dai_fmt,
-+		.set_clkdiv = wm8510_set_dai_clkdiv,
-+		.set_pll = wm8510_set_dai_pll,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8510_dai);
-+
-+static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8510_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8510_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8510 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8510_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "WM8510";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8510_read_reg_cache;
-+	codec->write = wm8510_write;
-+	codec->dapm_event = wm8510_dapm_event;
-+	codec->dai = &wm8510_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8510_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8510_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8510_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8510_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8510_reg);
-+
-+	wm8510_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "wm8510: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8510_add_controls(codec);
-+	wm8510_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8510: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8510_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8510 2 wire address is 0x1a
-+ */
-+#define I2C_DRIVERID_WM8510 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8510_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8510_socdev;
-+	struct wm8510_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if(ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8510_init(socdev);
-+	if(ret < 0) {
-+		err("failed to initialise WM8510\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int wm8510_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8510_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8510_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8510_i2c_driver = {
-+	.driver = {
-+		.name = "WM8510 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8510,
-+	.attach_adapter = wm8510_i2c_attach,
-+	.detach_client =  wm8510_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8510",
-+	.driver = &wm8510_i2c_driver,
-+};
-+#endif
-+
-+static int wm8510_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8510_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("WM8510 Audio Codec %s", WM8510_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8510_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8510_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8510_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8510_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8510 = {
-+	.probe = 	wm8510_probe,
-+	.remove = 	wm8510_remove,
-+	.suspend = 	wm8510_suspend,
-+	.resume =	wm8510_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
-+
-+MODULE_DESCRIPTION("ASoC WM8510 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8510.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8510.h
-@@ -0,0 +1,103 @@
-+/*
-+ * wm8510.h  --  WM8510 Soc Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8510_H
-+#define _WM8510_H
-+
-+/* WM8510 register space */
-+
-+#define WM8510_RESET		0x0
-+#define WM8510_POWER1		0x1
-+#define WM8510_POWER2		0x2
-+#define WM8510_POWER3		0x3
-+#define WM8510_IFACE		0x4
-+#define WM8510_COMP			0x5
-+#define WM8510_CLOCK		0x6
-+#define WM8510_ADD			0x7
-+#define WM8510_GPIO			0x8
-+#define WM8510_DAC			0xa
-+#define WM8510_DACVOL		0xb
-+#define WM8510_ADC			0xe
-+#define WM8510_ADCVOL		0xf
-+#define WM8510_EQ1			0x12
-+#define WM8510_EQ2			0x13
-+#define WM8510_EQ3			0x14
-+#define WM8510_EQ4			0x15
-+#define WM8510_EQ5			0x16
-+#define WM8510_DACLIM1		0x18
-+#define WM8510_DACLIM2		0x19
-+#define WM8510_NOTCH1		0x1b
-+#define WM8510_NOTCH2		0x1c
-+#define WM8510_NOTCH3		0x1d
-+#define WM8510_NOTCH4		0x1e
-+#define WM8510_ALC1			0x20
-+#define WM8510_ALC2			0x21
-+#define WM8510_ALC3			0x22
-+#define WM8510_NGATE		0x23
-+#define WM8510_PLLN			0x24
-+#define WM8510_PLLK1		0x25
-+#define WM8510_PLLK2		0x26
-+#define WM8510_PLLK3		0x27
-+#define WM8510_ATTEN		0x28
-+#define WM8510_INPUT		0x2c
-+#define WM8510_INPPGA		0x2d
-+#define WM8510_ADCBOOST		0x2f
-+#define WM8510_OUTPUT		0x31
-+#define WM8510_SPKMIX		0x32
-+#define WM8510_SPKVOL		0x36
-+#define WM8510_MONOMIX		0x38
-+
-+#define WM8510_CACHEREGNUM 	57
-+
-+/* Clock divider Id's */
-+#define WM8510_OPCLKDIV		0
-+#define WM8510_MCLKDIV		1
-+#define WM8510_ADCCLK		2
-+#define WM8510_DACCLK		3
-+#define WM8510_BCLKDIV		4
-+
-+/* DAC clock dividers */
-+#define WM8510_DACCLK_F2	(1 << 3)
-+#define WM8510_DACCLK_F4	(0 << 3)
-+
-+/* ADC clock dividers */
-+#define WM8510_ADCCLK_F2	(1 << 3)
-+#define WM8510_ADCCLK_F4	(0 << 3)
-+
-+/* PLL Out dividers */
-+#define WM8510_OPCLKDIV_1	(0 << 4)
-+#define WM8510_OPCLKDIV_2	(1 << 4)
-+#define WM8510_OPCLKDIV_3	(2 << 4)
-+#define WM8510_OPCLKDIV_4	(3 << 4)
-+
-+/* BCLK clock dividers */
-+#define WM8510_BCLKDIV_1	(0 << 2)
-+#define WM8510_BCLKDIV_2	(1 << 2)
-+#define WM8510_BCLKDIV_4	(2 << 2)
-+#define WM8510_BCLKDIV_8	(3 << 2)
-+#define WM8510_BCLKDIV_16	(4 << 2)
-+#define WM8510_BCLKDIV_32	(5 << 2)
-+
-+/* MCLK clock dividers */
-+#define WM8510_MCLKDIV_1	(0 << 5)
-+#define WM8510_MCLKDIV_1_5	(1 << 5)
-+#define WM8510_MCLKDIV_2	(2 << 5)
-+#define WM8510_MCLKDIV_3	(3 << 5)
-+#define WM8510_MCLKDIV_4	(4 << 5)
-+#define WM8510_MCLKDIV_6	(5 << 5)
-+#define WM8510_MCLKDIV_8	(6 << 5)
-+#define WM8510_MCLKDIV_12	(7 << 5)
-+
-+struct wm8510_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8510_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8510;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/imx/imx-ac97.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx-ac97.c
-@@ -0,0 +1,222 @@
-+/*
-+ * imx-ssi.c  --  SSI driver for Freescale IMX
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  Based on mxc-alsa-mc13783 (C) 2006 Freescale.
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    29th Aug 2006   Initial version.
-+ *
-+ */
-+
-+
-+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = {
-+	.name			= "SSI1 PCM Stereo out",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = emi_2_per,
-+		.watermark_level = SDMA_TXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI1_STX0,
-+		.event_id = DMA_REQ_SSI1_TX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = {
-+	.name			= "SSI1 PCM Stereo in",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_RXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI1_SRX0,
-+		.event_id = DMA_REQ_SSI1_RX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = {
-+	.name			= "SSI2 PCM Stereo out",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_TXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI2_STX0,
-+		.event_id = DMA_REQ_SSI2_TX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = {
-+	.name			= "SSI2 PCM Stereo in",
-+	.params = {
-+		.bd_number = 1,
-+		.transfer_type = per_2_emi,
-+		.watermark_level = SDMA_RXFIFO_WATERMARK,
-+		.word_size = TRANSFER_16BIT, // maybe add this in setup func
-+		.per_address = SSI2_SRX0,
-+		.event_id = DMA_REQ_SSI2_RX1,
-+		.peripheral_type = SSI,
-+	},
-+};
-+
-+static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
-+{
-+}
-+
-+static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
-+{
-+}
-+
-+static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
-+{
-+}
-+
-+static void imx_ssi_ac97_cold_reset(struct snd_ac97 *ac97)
-+{
-+}
-+
-+struct snd_ac97_bus_ops soc_ac97_ops = {
-+	.read	= imx_ssi_ac97_read,
-+	.write	= imx_ssi_ac97_write,
-+	.warm_reset	= imx_ssi_ac97_warm_reset,
-+	.reset	= imx_ssi_ac97_cold_reset,
-+};
-+
-+
-+static intimx_ssi1_ac97_probe(struct platform_device *pdev)
-+{
-+	int ret;
-+
-+
-+	return ret;
-+}
-+
-+static void imx_ssi1_ac97_remove(struct platform_device *pdev)
-+{
-+	/* shutdown SSI */
-+		if(rtd->cpu_dai->id == 0)
-+			SSI1_SCR &= ~SSI_SCR_SSIEN;
-+		else
-+			SSI2_SCR &= ~SSI_SCR_SSIEN;
-+	}
-+
-+}
-+
-+static int imx_ssi1_ac97_prepare(struct snd_pcm_substream *substream)
-+{
-+	// set vra
-+}
-+
-+static int imx_ssi_startup(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+
-+	if (!rtd->cpu_dai->active) {
-+
-+	}
-+
-+	return 0;
-+}
-+
-+static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+
-+	return ret;
-+}
-+
-+static void imx_ssi_shutdown(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+
-+
-+}
-+
-+#ifdef CONFIG_PM
-+static int imx_ssi_suspend(struct platform_device *dev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if(!dai->active)
-+		return 0;
-+
-+
-+	return 0;
-+}
-+
-+static int imx_ssi_resume(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+	if(!dai->active)
-+		return 0;
-+
-+	return 0;
-+}
-+
-+#else
-+#define imx_ssi_suspend	NULL
-+#define imx_ssi_resume	NULL
-+#endif
-+
-+#define IMX_AC97_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+	SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-+	SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000)
-+
-+struct snd_soc_cpu_dai imx_ssi_ac97_dai = {
-+	.name = "imx-ac97-1",
-+	.id = 0,
-+	.type = SND_SOC_DAI_AC97,
-+	.suspend = imx_ssi_suspend,
-+	.resume = imx_ssi_resume,
-+	.playback = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = IMX_AC97_RATES,},
-+	.capture = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = IMX_AC97_RATES,},
-+	.ops = {
-+		.probe = imx_ac97_probe,
-+		.remove = imx_ac97_shutdown,
-+		.trigger = imx_ssi_trigger,
-+		.prepare = imx_ssi_ac97_prepare,},
-+},
-+{
-+	.name = "imx-ac97-2",
-+	.id = 1,
-+	.type = SND_SOC_DAI_AC97,
-+	.suspend = imx_ssi_suspend,
-+	.resume = imx_ssi_resume,
-+	.playback = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = IMX_AC97_RATES,},
-+	.capture = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = IMX_AC97_RATES,},
-+	.ops = {
-+		.probe = imx_ac97_probe,
-+		.remove = imx_ac97_shutdown,
-+		.trigger = imx_ssi_trigger,
-+		.prepare = imx_ssi_ac97_prepare,},
-+};
-+
-+EXPORT_SYMBOL_GPL(imx_ssi_ac97_dai);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("i.MX ASoC AC97 driver");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8976.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8976.c
-@@ -0,0 +1,885 @@
-+/*
-+ * wm8976.c  --  WM8976 ALSA Soc Audio driver
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/version.h>
-+#include <linux/kernel.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8976.h"
-+
-+#define AUDIO_NAME "wm8976"
-+#define WM8976_VERSION "0.4"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8976_DEBUG 0
-+
-+#ifdef WM8976_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8976;
-+
-+/*
-+ * wm8976 register cache
-+ * We can't read the WM8976 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8976_reg[WM8976_CACHEREGNUM] = {
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0050, 0x0000, 0x0140, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x00ff,
-+    0x00ff, 0x0000, 0x0100, 0x00ff,
-+    0x00ff, 0x0000, 0x012c, 0x002c,
-+    0x002c, 0x002c, 0x002c, 0x0000,
-+    0x0032, 0x0000, 0x0000, 0x0000,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0038, 0x000b, 0x0032, 0x0000,
-+    0x0008, 0x000c, 0x0093, 0x00e9,
-+    0x0000, 0x0000, 0x0000, 0x0000,
-+    0x0033, 0x0010, 0x0010, 0x0100,
-+    0x0100, 0x0002, 0x0001, 0x0001,
-+    0x0039, 0x0039, 0x0039, 0x0039,
-+    0x0001, 0x0001,
-+};
-+
-+/*
-+ * read wm8976 register cache
-+ */
-+static inline unsigned int wm8976_read_reg_cache(struct snd_soc_codec  *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8976_RESET)
-+		return 0;
-+	if (reg >= WM8976_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8976 register cache
-+ */
-+static inline void wm8976_write_reg_cache(struct snd_soc_codec  *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8976_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8976 register space
-+ */
-+static int wm8976_write(struct snd_soc_codec  *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8976 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8976_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -1;
-+}
-+
-+#define wm8976_reset(c)	wm8976_write(c, WM8976_RESET, 0)
-+
-+static const char *wm8976_companding[] = {"Off", "NC", "u-law", "A-law" };
-+static const char *wm8976_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
-+static const char *wm8976_eqmode[] = {"Capture", "Playback" };
-+static const char *wm8976_bw[] = {"Narrow", "Wide" };
-+static const char *wm8976_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
-+static const char *wm8976_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
-+static const char *wm8976_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
-+static const char *wm8976_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
-+static const char *wm8976_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
-+static const char *wm8976_alc[] =
-+    {"ALC both on", "ALC left only", "ALC right only", "Limiter" };
-+
-+static const struct soc_enum wm8976_enum[] = {
-+	SOC_ENUM_SINGLE(WM8976_COMP, 1, 4, wm8976_companding), /* adc */
-+	SOC_ENUM_SINGLE(WM8976_COMP, 3, 4, wm8976_companding), /* dac */
-+	SOC_ENUM_SINGLE(WM8976_DAC,  4, 4, wm8976_deemp),
-+	SOC_ENUM_SINGLE(WM8976_EQ1,  8, 2, wm8976_eqmode),
-+
-+	SOC_ENUM_SINGLE(WM8976_EQ1,  5, 4, wm8976_eq1),
-+	SOC_ENUM_SINGLE(WM8976_EQ2,  8, 2, wm8976_bw),
-+	SOC_ENUM_SINGLE(WM8976_EQ2,  5, 4, wm8976_eq2),
-+	SOC_ENUM_SINGLE(WM8976_EQ3,  8, 2, wm8976_bw),
-+
-+	SOC_ENUM_SINGLE(WM8976_EQ3,  5, 4, wm8976_eq3),
-+	SOC_ENUM_SINGLE(WM8976_EQ4,  8, 2, wm8976_bw),
-+	SOC_ENUM_SINGLE(WM8976_EQ4,  5, 4, wm8976_eq4),
-+	SOC_ENUM_SINGLE(WM8976_EQ5,  8, 2, wm8976_bw),
-+
-+	SOC_ENUM_SINGLE(WM8976_EQ5,  5, 4, wm8976_eq5),
-+	SOC_ENUM_SINGLE(WM8976_ALC3,  8, 2, wm8976_alc),
-+};
-+
-+static const struct snd_kcontrol_new wm8976_snd_controls[] = {
-+SOC_SINGLE("Digital Loopback Switch", WM8976_COMP, 0, 1, 0),
-+
-+SOC_ENUM("ADC Companding", wm8976_enum[0]),
-+SOC_ENUM("DAC Companding", wm8976_enum[1]),
-+
-+SOC_SINGLE("Jack Detection Enable", WM8976_JACK1, 6, 1, 0),
-+
-+SOC_DOUBLE("DAC Inversion Switch", WM8976_DAC, 0, 1, 1, 0),
-+
-+SOC_DOUBLE_R("Headphone Playback Volume", WM8976_DACVOLL, WM8976_DACVOLR, 0, 127, 0),
-+
-+SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0),
-+SOC_SINGLE("High Pass Cut Off", WM8976_ADC, 4, 7, 0),
-+
-+SOC_DOUBLE("ADC Inversion Switch", WM8976_ADC, 0, 1, 1, 0),
-+
-+SOC_SINGLE("Capture Volume", WM8976_ADCVOL,  0, 127, 0),
-+
-+SOC_ENUM("Equaliser Function", wm8976_enum[3]),
-+SOC_ENUM("EQ1 Cut Off", wm8976_enum[4]),
-+SOC_SINGLE("EQ1 Volume", WM8976_EQ1,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ2 Bandwith", wm8976_enum[5]),
-+SOC_ENUM("EQ2 Cut Off", wm8976_enum[6]),
-+SOC_SINGLE("EQ2 Volume", WM8976_EQ2,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ3 Bandwith", wm8976_enum[7]),
-+SOC_ENUM("EQ3 Cut Off", wm8976_enum[8]),
-+SOC_SINGLE("EQ3 Volume", WM8976_EQ3,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ4 Bandwith", wm8976_enum[9]),
-+SOC_ENUM("EQ4 Cut Off", wm8976_enum[10]),
-+SOC_SINGLE("EQ4 Volume", WM8976_EQ4,  0, 31, 1),
-+
-+SOC_ENUM("Equaliser EQ5 Bandwith", wm8976_enum[11]),
-+SOC_ENUM("EQ5 Cut Off", wm8976_enum[12]),
-+SOC_SINGLE("EQ5 Volume", WM8976_EQ5,  0, 31, 1),
-+
-+SOC_SINGLE("DAC Playback Limiter Switch", WM8976_DACLIM1,  8, 1, 0),
-+SOC_SINGLE("DAC Playback Limiter Decay", WM8976_DACLIM1,  4, 15, 0),
-+SOC_SINGLE("DAC Playback Limiter Attack", WM8976_DACLIM1,  0, 15, 0),
-+
-+SOC_SINGLE("DAC Playback Limiter Threshold", WM8976_DACLIM2,  4, 7, 0),
-+SOC_SINGLE("DAC Playback Limiter Boost", WM8976_DACLIM2,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Enable Switch", WM8976_ALC1,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Max Gain", WM8976_ALC1,  3, 7, 0),
-+SOC_SINGLE("ALC Capture Min Gain", WM8976_ALC1,  0, 7, 0),
-+
-+SOC_SINGLE("ALC Capture ZC Switch", WM8976_ALC2,  8, 1, 0),
-+SOC_SINGLE("ALC Capture Hold", WM8976_ALC2,  4, 7, 0),
-+SOC_SINGLE("ALC Capture Target", WM8976_ALC2,  0, 15, 0),
-+
-+SOC_ENUM("ALC Capture Mode", wm8976_enum[13]),
-+SOC_SINGLE("ALC Capture Decay", WM8976_ALC3,  4, 15, 0),
-+SOC_SINGLE("ALC Capture Attack", WM8976_ALC3,  0, 15, 0),
-+
-+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8976_NGATE,  3, 1, 0),
-+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8976_NGATE,  0, 7, 0),
-+
-+SOC_SINGLE("Capture PGA ZC Switch", WM8976_INPPGA,  7, 1, 0),
-+SOC_SINGLE("Capture PGA Volume", WM8976_INPPGA,  0, 63, 0),
-+
-+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8976_HPVOLL,  WM8976_HPVOLR, 7, 1, 0),
-+SOC_DOUBLE_R("Headphone Playback Switch", WM8976_HPVOLL,  WM8976_HPVOLR, 6, 1, 1),
-+SOC_DOUBLE_R("Headphone Playback Volume", WM8976_HPVOLL,  WM8976_HPVOLR, 0, 63, 0),
-+
-+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8976_SPKVOLL,  WM8976_SPKVOLR, 7, 1, 0),
-+SOC_DOUBLE_R("Speaker Playback Switch", WM8976_SPKVOLL,  WM8976_SPKVOLR, 6, 1, 1),
-+SOC_DOUBLE_R("Speaker Playback Volume", WM8976_SPKVOLL,  WM8976_SPKVOLR, 0, 63, 0),
-+
-+SOC_SINGLE("Capture Boost(+20dB)", WM8976_ADCBOOST, 8, 1, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8976_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8976_snd_controls); i++) {
-+		err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8976_snd_controls[i],codec, NULL));
-+		if (err < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Left Output Mixer */
-+static const snd_kcontrol_new_t wm8976_left_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_OUTPUT, 6, 1, 1),
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_MIXL, 0, 1, 1),
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXL, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXL, 5, 1, 0),
-+};
-+
-+/* Right Output Mixer */
-+static const snd_kcontrol_new_t wm8976_right_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_OUTPUT, 5, 1, 1),
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_MIXR, 0, 1, 1),
-+SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXR, 1, 1, 0),
-+SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXR, 5, 1, 0),
-+};
-+
-+/* Left AUX Input boost vol */
-+static const snd_kcontrol_new_t wm8976_laux_boost_controls =
-+SOC_DAPM_SINGLE("Aux Volume", WM8976_ADCBOOST, 0, 3, 0);
-+
-+/* Left Input boost vol */
-+static const snd_kcontrol_new_t wm8976_lmic_boost_controls =
-+SOC_DAPM_SINGLE("Input Volume", WM8976_ADCBOOST, 4, 3, 0);
-+
-+/* Left Aux In to PGA */
-+static const snd_kcontrol_new_t wm8976_laux_capture_boost_controls =
-+SOC_DAPM_SINGLE("Capture Switch", WM8976_ADCBOOST,  8, 1, 0);
-+
-+/* Left Input P In to PGA */
-+static const snd_kcontrol_new_t wm8976_lmicp_capture_boost_controls =
-+SOC_DAPM_SINGLE("Input P Capture Boost Switch", WM8976_INPUT,  0, 1, 0);
-+
-+/* Left Input N In to PGA */
-+static const snd_kcontrol_new_t wm8976_lmicn_capture_boost_controls =
-+SOC_DAPM_SINGLE("Input N Capture Boost Switch", WM8976_INPUT,  1, 1, 0);
-+
-+// TODO Widgets
-+static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = {
-+#if 0
-+//SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0),
-+//SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0),
-+
-+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0,
-+	&wm8976_speaker_mixer_controls[0],
-+	ARRAY_SIZE(wm8976_speaker_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0,
-+	&wm8976_mono_mixer_controls[0],
-+	ARRAY_SIZE(wm8976_mono_mixer_controls)),
-+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0),
-+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0),
-+SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0),
-+SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0),
-+
-+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8976_aux_boost_controls, 1),
-+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8976_mic_boost_controls, 1),
-+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0,
-+	&wm8976_capture_boost_controls),
-+
-+SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0),
-+
-+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0),
-+
-+SND_SOC_DAPM_INPUT("MICN"),
-+SND_SOC_DAPM_INPUT("MICP"),
-+SND_SOC_DAPM_INPUT("AUX"),
-+SND_SOC_DAPM_OUTPUT("MONOOUT"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
-+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
-+#endif
-+};
-+
-+static const char *audio_map[][3] = {
-+	/* Mono output mixer */
-+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
-+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Speaker output mixer */
-+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
-+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
-+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
-+
-+	/* Outputs */
-+	{"Mono Out", NULL, "Mono Mixer"},
-+	{"MONOOUT", NULL, "Mono Out"},
-+	{"SpkN Out", NULL, "Speaker Mixer"},
-+	{"SpkP Out", NULL, "Speaker Mixer"},
-+	{"SPKOUTN", NULL, "SpkN Out"},
-+	{"SPKOUTP", NULL, "SpkP Out"},
-+
-+	/* Boost Mixer */
-+	{"Boost Mixer", NULL, "ADC"},
-+	{"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"},
-+	{"Aux Boost", "Aux Volume", "Boost Mixer"},
-+	{"Capture Boost", "Capture Switch", "Boost Mixer"},
-+	{"Mic Boost", "Mic Volume", "Boost Mixer"},
-+
-+	/* Inputs */
-+	{"MICP", NULL, "Mic Boost"},
-+	{"MICN", NULL, "Mic PGA"},
-+	{"Mic PGA", NULL, "Capture Boost"},
-+	{"AUX", NULL, "Aux Input"},
-+
-+    /*  */
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8976_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path map */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1],
-+            audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+struct pll_ {
-+	unsigned int in_hz, out_hz;
-+	unsigned int pre:4; /* prescale - 1 */
-+	unsigned int n:4;
-+	unsigned int k;
-+};
-+
-+struct pll_ pll[] = {
-+	{12000000, 11289600, 0, 7, 0x86c220},
-+	{12000000, 12288000, 0, 8, 0x3126e8},
-+	{13000000, 11289600, 0, 6, 0xf28bd4},
-+	{13000000, 12288000, 0, 7, 0x8fd525},
-+	{12288000, 11289600, 0, 7, 0x59999a},
-+	{11289600, 12288000, 0, 8, 0x80dee9},
-+	/* TODO: liam - add more entries */
-+};
-+
-+static int wm8976_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	int i;
-+	u16 reg;
-+
-+	if(freq_in == 0 || freq_out == 0) {
-+		reg = wm8976_read_reg_cache(codec, WM8976_POWER1);
-+		wm8976_write(codec, WM8976_POWER1, reg & 0x1df);
-+		return 0;
-+	}
-+
-+	for(i = 0; i < ARRAY_SIZE(pll); i++) {
-+		if (freq_in == pll[i].in_hz && freq_out == pll[i].out_hz) {
-+			wm8976_write(codec, WM8976_PLLN, (pll[i].pre << 4) | pll[i].n);
-+			wm8976_write(codec, WM8976_PLLK1, pll[i].k >> 18);
-+			wm8976_write(codec, WM8976_PLLK1, (pll[i].k >> 9) && 0x1ff);
-+			wm8976_write(codec, WM8976_PLLK1, pll[i].k && 0x1ff);
-+			reg = wm8976_read_reg_cache(codec, WM8976_POWER1);
-+			wm8976_write(codec, WM8976_POWER1, reg | 0x020);
-+			return 0;
-+		}
-+	}
-+	return -EINVAL;
-+}
-+
-+static int wm8976_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0x3;
-+	u16 clk = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0xfffe;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		clk |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0010;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0008;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x00018;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0180;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0100;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0080;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	wm8976_write(codec, WM8976_IFACE, iface);
-+	return 0;
-+}
-+
-+static int wm8976_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 iface = wm8976_read_reg_cache(codec, WM8976_IFACE) & 0xff9f;
-+	u16 adn = wm8976_read_reg_cache(codec, WM8976_ADD) & 0x1f1;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0020;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0040;
-+		break;
-+	}
-+
-+	/* filter coefficient */
-+	switch (params_rate(params)) {
-+	case SNDRV_PCM_RATE_8000:
-+		adn |= 0x5 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_11025:
-+		adn |= 0x4 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_16000:
-+		adn |= 0x3 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_22050:
-+		adn |= 0x2 << 1;
-+		break;
-+	case SNDRV_PCM_RATE_32000:
-+		adn |= 0x1 << 1;
-+		break;
-+	}
-+
-+	/* set iface */
-+	wm8976_write(codec, WM8976_IFACE, iface);
-+	wm8976_write(codec, WM8976_ADD, adn);
-+	return 0;
-+}
-+
-+static int wm8976_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8976_MCLKDIV:
-+		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x11f;
-+		wm8976_write(codec, WM8976_CLOCK, reg | div);
-+		break;
-+	case WM8976_BCLKDIV:
-+		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x1c7;
-+		wm8976_write(codec, WM8976_CLOCK, reg | div);
-+		break;
-+	case WM8976_OPCLKDIV:
-+		reg = wm8976_read_reg_cache(codec, WM8976_GPIO) & 0x1cf;
-+		wm8976_write(codec, WM8976_GPIO, reg | div);
-+		break;
-+	case WM8976_DACOSR:
-+		reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0x1f7;
-+		wm8976_write(codec, WM8976_DAC, reg | div);
-+		break;
-+	case WM8976_ADCOSR:
-+		reg = wm8976_read_reg_cache(codec, WM8976_ADC) & 0x1f7;
-+		wm8976_write(codec, WM8976_ADC, reg | div);
-+		break;
-+	case WM8976_MCLKSEL:
-+		reg = wm8976_read_reg_cache(codec, WM8976_CLOCK) & 0x0ff;
-+		wm8976_write(codec, WM8976_CLOCK, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+static int wm8976_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0xffbf;
-+
-+	if(mute)
-+		wm8976_write(codec, WM8976_DAC, mute_reg | 0x40);
-+	else
-+		wm8976_write(codec, WM8976_DAC, mute_reg);
-+
-+	return 0;
-+}
-+
-+/* TODO: liam need to make this lower power with dapm */
-+static int wm8976_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, clk and osc on, dac unmute, active */
-+		wm8976_write(codec, WM8976_POWER1, 0x1ff);
-+		wm8976_write(codec, WM8976_POWER2, 0x1ff);
-+		wm8976_write(codec, WM8976_POWER3, 0x1ff);
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, dac mute, inactive */
-+
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		wm8976_write(codec, WM8976_POWER1, 0x0);
-+		wm8976_write(codec, WM8976_POWER2, 0x0);
-+		wm8976_write(codec, WM8976_POWER3, 0x0);
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+#define WM8976_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000)
-+
-+#define WM8976_FORMATS \
-+	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
-+	SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8976_dai = {
-+	.name = "WM8976 HiFi",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8976_RATES,
-+		.formats = WM8976_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 1,
-+		.rates = WM8976_RATES,
-+		.formats = WM8976_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8976_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8976_mute,
-+		.set_fmt = wm8976_set_dai_fmt,
-+		.set_clkdiv = wm8976_set_dai_clkdiv,
-+		.set_pll = wm8976_set_dai_pll,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8976_dai);
-+
-+static int wm8976_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8976_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8976_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8976_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8976 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8976_init(struct snd_soc_device* socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int ret = 0;
-+
-+	codec->name = "WM8976";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8976_read_reg_cache;
-+	codec->write = wm8976_write;
-+	codec->dapm_event = wm8976_dapm_event;
-+	codec->dai = &wm8976_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8976_reg);
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8976_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache, wm8976_reg,
-+		sizeof(u16) * ARRAY_SIZE(wm8976_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8976_reg);
-+
-+	wm8976_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if(ret < 0) {
-+		printk(KERN_ERR "wm8976: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8976_add_controls(codec);
-+	wm8976_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8976: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8976_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8976 2 wire address is 0x1a
-+ */
-+#define I2C_DRIVERID_WM8976 0xfefe /* liam -  need a proper id */
-+
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8976_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8976_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8976_socdev;
-+	struct wm8976_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL){
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+
-+	i2c_set_clientdata(i2c, codec);
-+
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if(ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8976_init(socdev);
-+	if(ret < 0) {
-+		err("failed to initialise WM8976\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+
-+}
-+
-+static int wm8976_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8976_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8976_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8976_i2c_driver = {
-+	.driver = {
-+		.name = "WM8976 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8976,
-+	.attach_adapter = wm8976_i2c_attach,
-+	.detach_client =  wm8976_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8976",
-+	.driver = &wm8976_i2c_driver,
-+};
-+#endif
-+
-+static int wm8976_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8976_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("WM8976 Audio Codec %s", WM8976_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8976_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8976_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8976_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8976_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8976 = {
-+	.probe = 	wm8976_probe,
-+	.remove = 	wm8976_remove,
-+	.suspend = 	wm8976_suspend,
-+	.resume =	wm8976_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8976);
-+
-+MODULE_DESCRIPTION("ASoC WM8976 driver");
-+MODULE_AUTHOR("Graeme Gregory");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8976.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8976.h
-@@ -0,0 +1,112 @@
-+/*
-+ * wm8976.h  --  WM8976 Soc Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8976_H
-+#define _WM8976_H
-+
-+/* WM8976 register space */
-+
-+#define WM8976_RESET		0x0
-+#define WM8976_POWER1		0x1
-+#define WM8976_POWER2		0x2
-+#define WM8976_POWER3		0x3
-+#define WM8976_IFACE		0x4
-+#define WM8976_COMP			0x5
-+#define WM8976_CLOCK		0x6
-+#define WM8976_ADD			0x7
-+#define WM8976_GPIO			0x8
-+#define WM8976_JACK1        0x9
-+#define WM8976_DAC			0xa
-+#define WM8976_DACVOLL	    0xb
-+#define WM8976_DACVOLR      0xc
-+#define WM8976_JACK2        0xd
-+#define WM8976_ADC			0xe
-+#define WM8976_ADCVOL		0xf
-+#define WM8976_EQ1			0x12
-+#define WM8976_EQ2			0x13
-+#define WM8976_EQ3			0x14
-+#define WM8976_EQ4			0x15
-+#define WM8976_EQ5			0x16
-+#define WM8976_DACLIM1		0x18
-+#define WM8976_DACLIM2		0x19
-+#define WM8976_NOTCH1		0x1b
-+#define WM8976_NOTCH2		0x1c
-+#define WM8976_NOTCH3		0x1d
-+#define WM8976_NOTCH4		0x1e
-+#define WM8976_ALC1			0x20
-+#define WM8976_ALC2			0x21
-+#define WM8976_ALC3			0x22
-+#define WM8976_NGATE		0x23
-+#define WM8976_PLLN			0x24
-+#define WM8976_PLLK1		0x25
-+#define WM8976_PLLK2		0x26
-+#define WM8976_PLLK3		0x27
-+#define WM8976_3D           0x29
-+#define WM8976_BEEP         0x2b
-+#define WM8976_INPUT		0x2c
-+#define WM8976_INPPGA	  	0x2d
-+#define WM8976_ADCBOOST		0x2f
-+#define WM8976_OUTPUT		0x31
-+#define WM8976_MIXL	        0x32
-+#define WM8976_MIXR         0x33
-+#define WM8976_HPVOLL		0x34
-+#define WM8976_HPVOLR       0x35
-+#define WM8976_SPKVOLL      0x36
-+#define WM8976_SPKVOLR      0x37
-+#define WM8976_OUT3MIX		0x38
-+#define WM8976_MONOMIX      0x39
-+
-+#define WM8976_CACHEREGNUM 	58
-+
-+/*
-+ * WM8976 Clock dividers
-+ */
-+#define WM8976_MCLKDIV 		0
-+#define WM8976_BCLKDIV		1
-+#define WM8976_OPCLKDIV		2
-+#define WM8976_DACOSR		3
-+#define WM8976_ADCOSR		4
-+#define WM8976_MCLKSEL		5
-+
-+#define WM8976_MCLK_MCLK		(0 << 8)
-+#define WM8976_MCLK_PLL			(1 << 8)
-+
-+#define WM8976_MCLK_DIV_1		(0 << 5)
-+#define WM8976_MCLK_DIV_1_5		(1 << 5)
-+#define WM8976_MCLK_DIV_2		(2 << 5)
-+#define WM8976_MCLK_DIV_3		(3 << 5)
-+#define WM8976_MCLK_DIV_4		(4 << 5)
-+#define WM8976_MCLK_DIV_5_5		(5 << 5)
-+#define WM8976_MCLK_DIV_6		(6 << 5)
-+
-+#define WM8976_BCLK_DIV_1		(0 << 2)
-+#define WM8976_BCLK_DIV_2		(1 << 2)
-+#define WM8976_BCLK_DIV_4		(2 << 2)
-+#define WM8976_BCLK_DIV_8		(3 << 2)
-+#define WM8976_BCLK_DIV_16		(4 << 2)
-+#define WM8976_BCLK_DIV_32		(5 << 2)
-+
-+#define WM8976_DACOSR_64		(0 << 3)
-+#define WM8976_DACOSR_128		(1 << 3)
-+
-+#define WM8976_ADCOSR_64		(0 << 3)
-+#define WM8976_ADCOSR_128		(1 << 3)
-+
-+#define WM8976_OPCLK_DIV_1		(0 << 4)
-+#define WM8976_OPCLK_DIV_2		(1 << 4)
-+#define WM8976_OPCLK_DIV_3		(2 << 4)
-+#define WM8976_OPCLK_DIV_4		(3 << 4)
-+
-+struct wm8976_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8976_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8976;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/imx/imx21-pcm.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx21-pcm.c
-@@ -0,0 +1,454 @@
-+/*
-+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * Based on pxa2xx-pcm.c by	Nicolas Pitre, (C) 2004 MontaVista Software, Inc.
-+ * and on mxc-alsa-mc13783 (C) 2006 Freescale.
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ *
-+ *  Revision history
-+ *    29th Aug 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/platform_device.h>
-+#include <linux/slab.h>
-+#include <linux/dma-mapping.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <asm/dma.h>
-+#include <asm/hardware.h>
-+
-+#include "imx-pcm.h"
-+
-+/* debug */
-+#define IMX_DEBUG 0
-+#if IMX_DEBUG
-+#define dbg(format, arg...) printk(format, ## arg)
-+#else
-+#define dbg(format, arg...)
-+#endif
-+
-+static const struct snd_pcm_hardware mxc_pcm_hardware = {
-+	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
-+				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-+				   SNDRV_PCM_INFO_MMAP |
-+				   SNDRV_PCM_INFO_MMAP_VALID |
-+				   SNDRV_PCM_INFO_PAUSE |
-+				   SNDRV_PCM_INFO_RESUME),
-+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-+					SNDRV_PCM_FMTBIT_S24_LE,
-+	.buffer_bytes_max	= 32 * 1024,
-+	.period_bytes_min	= 64,
-+	.period_bytes_max	= 8 * 1024,
-+	.periods_min		= 2,
-+	.periods_max		= 255,
-+	.fifo_size		= 0,
-+};
-+
-+struct mxc_runtime_data {
-+	int dma_ch;
-+	struct mxc_pcm_dma_param *dma_params;
-+};
-+
-+/*!
-+  * This function stops the current dma transfert for playback
-+  * and clears the dma pointers.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  */
-+static void audio_stop_dma(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	unsigned int offset  dma_size * s->periods;
-+	unsigned long flags;
-+
-+	spin_lock_irqsave(&prtd->dma_lock, flags);
-+
-+	dbg("MXC : audio_stop_dma active = 0\n");
-+	prtd->active = 0;
-+	prtd->period = 0;
-+	prtd->periods = 0;
-+
-+	/* this stops the dma channel and clears the buffer ptrs */
-+	mxc_dma_stop(prtd->dma_wchannel);
-+	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_TO_DEVICE);
-+	else
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_FROM_DEVICE);
-+
-+	spin_unlock_irqrestore(&prtd->dma_lock, flags);
-+}
-+
-+/*!
-+  * This function is called whenever a new audio block needs to be
-+  * transferred to mc13783. The function receives the address and the size
-+  * of the new block and start a new DMA transfer.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  */
-+static int dma_new_period(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime =  substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int dma_size;
-+	unsigned int offset;
-+	int ret=0;
-+	dma_request_t sdma_request;
-+
-+	if (prtd->active){
-+		memset(&sdma_request, 0, sizeof(dma_request_t));
-+		dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	    dbg("s->period (%x) runtime->periods (%d)\n",
-+			s->period,runtime->periods);
-+		dbg("runtime->period_size (%d) dma_size (%d)\n",
-+			(unsigned int)runtime->period_size,
-+			runtime->dma_bytes);
-+
-+       	offset = dma_size * prtd->period;
-+		snd_assert(dma_size <= DMA_BUF_SIZE, );
-+		if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+			sdma_request.sourceAddr = (char*)(dma_map_single(NULL,
-+				runtime->dma_area + offset, dma_size, DMA_TO_DEVICE));
-+		else
-+			sdma_request.destAddr = (char*)(dma_map_single(NULL,
-+				runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE));
-+		sdma_request.count = dma_size;
-+
-+		dbg("MXC: Start DMA offset (%d) size (%d)\n", offset,
-+						 runtime->dma_bytes);
-+
-+       	mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0);
-+		if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) {
-+			dbg("audio_process_dma: cannot queue DMA buffer\
-+							(%i)\n", ret);
-+			return err;
-+		}
-+		prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */
-+		prtd->period++;
-+		prtd->period %= runtime->periods;
-+    }
-+	return ret;
-+}
-+
-+
-+/*!
-+  * This is a callback which will be called
-+  * when a TX transfer finishes. The call occurs
-+  * in interrupt context.
-+  *
-+  * @param	dat	pointer to the structure of the current stream.
-+  *
-+  */
-+static void audio_dma_irq(void *data)
-+{
-+	struct snd_pcm_substream *substream;
-+	struct snd_pcm_runtime *runtime;
-+	struct mxc_runtime_data *prtd;
-+	unsigned int dma_size;
-+	unsigned int previous_period;
-+	unsigned int offset;
-+
-+	substream = data;
-+	runtime = substream->runtime;
-+	prtd = runtime->private_data;
-+	previous_period  = prtd->periods;
-+	dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	offset = dma_size * previous_period;
-+
-+	prtd->tx_spin = 0;
-+	prtd->periods++;
-+	prtd->periods %= runtime->periods;
-+
-+	/*
-+	  * Give back to the CPU the access to the non cached memory
-+	  */
-+	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_TO_DEVICE);
-+	else
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_FROM_DEVICE);
-+	/*
-+	  * If we are getting a callback for an active stream then we inform
-+	  * the PCM middle layer we've finished a period
-+	  */
-+ 	if (prtd->active)
-+		snd_pcm_period_elapsed(substream);
-+
-+	/*
-+	  * Trig next DMA transfer
-+	  */
-+	dma_new_period(substream);
-+}
-+
-+/*!
-+  * This function configures the hardware to allow audio
-+  * playback operations. It is called by ALSA framework.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  * @return              0 on success, -1 otherwise.
-+  */
-+static int
-+snd_mxc_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	int ret = 0;
-+	prtd->period = 0;
-+	prtd->periods = 0;
-+
-+	dma_channel_params params;
-+	int channel = 0; // passed in ?
-+
-+	if ((ret  = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){
-+		dbg("error requesting a write dma channel\n");
-+		return ret;
-+	}
-+
-+	/* configure DMA params */
-+	memset(&params, 0, sizeof(dma_channel_params));
-+	params.bd_number = 1;
-+	params.arg = s;
-+	params.callback = callback;
-+	params.transfer_type = emi_2_per;
-+	params.watermark_level = SDMA_TXFIFO_WATERMARK;
-+	params.word_size = TRANSFER_16BIT;
-+	//dbg(KERN_ERR "activating connection SSI1 - SDMA\n");
-+	params.per_address = SSI1_BASE_ADDR;
-+	params.event_id = DMA_REQ_SSI1_TX1;
-+	params.peripheral_type = SSI;
-+
-+	/* set up chn with params */
-+	mxc_dma_setup_channel(channel, &params);
-+	s->dma_wchannel = channel;
-+
-+	return ret;
-+}
-+
-+static int mxc_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	int ret;
-+
-+	if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
-+		return ret;
-+	runtime->dma_addr = virt_to_phys(runtime->dma_area);
-+
-+	return ret;
-+}
-+
-+static int mxc_pcm_hw_free(struct snd_pcm_substream *substream)
-+{
-+	return snd_pcm_lib_free_pages(substream);
-+}
-+
-+static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct mxc_runtime_data *prtd = substream->runtime->private_data;
-+	int ret = 0;
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+		prtd->tx_spin = 0;
-+		/* requested stream startup */
-+		prtd->active = 1;
-+        ret = dma_new_period(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+		/* requested stream shutdown */
-+		ret = audio_stop_dma(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+		prtd->active = 0;
-+		prtd->periods = 0;
-+		break;
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+		prtd->active = 1;
-+		prtd->tx_spin = 0;
-+		ret = dma_new_period(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		prtd->active = 0;
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		prtd->active = 1;
-+		if (prtd->old_offset) {
-+			prtd->tx_spin = 0;
-+            ret = dma_new_period(substream);
-+		}
-+		break;
-+	default:
-+		ret = -EINVAL;
-+		break;
-+	}
-+
-+	return ret;
-+}
-+
-+static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int offset = 0;
-+
-+	/* tx_spin value is used here to check if a transfert is active */
-+	if (prtd->tx_spin){
-+		offset = (runtime->period_size * (prtd->periods)) +
-+						(runtime->period_size >> 1);
-+		if (offset >= runtime->buffer_size)
-+			offset = runtime->period_size >> 1;
-+	} else {
-+		offset = (runtime->period_size * (s->periods));
-+		if (offset >= runtime->buffer_size)
-+			offset = 0;
-+	}
-+
-+	return offset;
-+}
-+
-+
-+static int mxc_pcm_open(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd;
-+	int ret;
-+
-+	snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware);
-+
-+	if ((err = snd_pcm_hw_constraint_integer(runtime,
-+					SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
-+		return err;
-+	if ((err = snd_pcm_hw_constraint_list(runtime, 0,
-+			SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0)
-+		return err;
-+	msleep(10); // liam - why
-+
-+	/* setup DMA controller for playback */
-+	if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK],
-+					audio_dma_irq)) < 0 )
-+		return err;
-+
-+	if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) {
-+		ret = -ENOMEM;
-+		goto out;
-+	}
-+
-+	runtime->private_data = prtd;
-+	return 0;
-+
-+ err1:
-+	kfree(prtd);
-+ out:
-+	return ret;
-+}
-+
-+static int mxc_pcm_close(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+
-+//	mxc_mc13783_t *chip;
-+	audio_stream_t *s;
-+	device_data_t* device;
-+	int ssi;
-+
-+	//chip = snd_pcm_substream_chip(substream);
-+	s = &chip->s[substream->pstr->stream];
-+	device = &s->stream_device;
-+	ssi = device->ssi;
-+
-+	//disable_stereodac();
-+
-+	ssi_transmit_enable(ssi, false);
-+	ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable);
-+	ssi_tx_fifo_enable(ssi, ssi_fifo_0, false);
-+	ssi_enable(ssi, false);
-+
-+	chip->s[substream->pstr->stream].stream = NULL;
-+
-+	return 0;
-+}
-+
-+static int
-+mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-+				     runtime->dma_area,
-+				     runtime->dma_addr,
-+				     runtime->dma_bytes);
-+}
-+
-+struct snd_pcm_ops mxc_pcm_ops = {
-+	.open		= mxc_pcm_open,
-+	.close		= mxc_pcm_close,
-+	.ioctl		= snd_pcm_lib_ioctl,
-+	.hw_params	= mxc_pcm_hw_params,
-+	.hw_free	= mxc_pcm_hw_free,
-+	.prepare	= mxc_pcm_prepare,
-+	.trigger	= mxc_pcm_trigger,
-+	.pointer	= mxc_pcm_pointer,
-+	.mmap		= mxc_pcm_mmap,
-+};
-+
-+static u64 mxc_pcm_dmamask = 0xffffffff;
-+
-+int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
-+	struct snd_pcm *pcm)
-+{
-+	int ret = 0;
-+
-+	if (!card->dev->dma_mask)
-+		card->dev->dma_mask = &mxc_pcm_dmamask;
-+	if (!card->dev->coherent_dma_mask)
-+		card->dev->coherent_dma_mask = 0xffffffff;
-+
-+	if (dai->playback.channels_min) {
-+		ret = mxc_pcm_preallocate_dma_buffer(pcm,
-+			SNDRV_PCM_STREAM_PLAYBACK);
-+		if (ret)
-+			goto out;
-+	}
-+
-+	if (dai->capture.channels_min) {
-+		ret = mxc_pcm_preallocate_dma_buffer(pcm,
-+			SNDRV_PCM_STREAM_CAPTURE);
-+		if (ret)
-+			goto out;
-+	}
-+ out:
-+	return ret;
-+}
-+
-+struct snd_soc_platform mxc_soc_platform = {
-+	.name		= "mxc-audio",
-+	.pcm_ops 	= &mxc_pcm_ops,
-+	.pcm_new	= mxc_pcm_new,
-+	.pcm_free	= mxc_pcm_free_dma_buffers,
-+};
-+
-+EXPORT_SYMBOL_GPL(mxc_soc_platform);
-+
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_DESCRIPTION("Freescale i.MX PCM DMA module");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/imx/imx21-pcm.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx21-pcm.h
-@@ -0,0 +1,237 @@
-+/*
-+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _MXC_PCM_H
-+#define _MXC_PCM_H
-+
-+struct {
-+	char *name;			/* stream identifier */
-+	dma_channel_params dma_params;
-+} mxc_pcm_dma_param;
-+
-+extern struct snd_soc_cpu_dai mxc_ssi_dai[3];
-+
-+/* platform data */
-+extern struct snd_soc_platform mxc_soc_platform;
-+extern struct snd_ac97_bus_ops mxc_ac97_ops;
-+
-+/* temp until imx-regs.h is up2date */
-+#define SSI1_STX0   __REG(IMX_SSI1_BASE + 0x00)
-+#define SSI1_STX0_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x00)
-+#define SSI1_STX1   __REG(IMX_SSI1_BASE + 0x04)
-+#define SSI1_STX1_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x04)
-+#define SSI1_SRX0   __REG(IMX_SSI1_BASE + 0x08)
-+#define SSI1_SRX0_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x08)
-+#define SSI1_SRX1   __REG(IMX_SSI1_BASE + 0x0c)
-+#define SSI1_SRX1_PHYS   __PHYS_REG(IMX_SSI1_BASE + 0x0c)
-+#define SSI1_SCR    __REG(IMX_SSI1_BASE + 0x10)
-+#define SSI1_SISR   __REG(IMX_SSI1_BASE + 0x14)
-+#define SSI1_SIER   __REG(IMX_SSI1_BASE + 0x18)
-+#define SSI1_STCR   __REG(IMX_SSI1_BASE + 0x1c)
-+#define SSI1_SRCR   __REG(IMX_SSI1_BASE + 0x20)
-+#define SSI1_STCCR  __REG(IMX_SSI1_BASE + 0x24)
-+#define SSI1_SRCCR  __REG(IMX_SSI1_BASE + 0x28)
-+#define SSI1_SFCSR  __REG(IMX_SSI1_BASE + 0x2c)
-+#define SSI1_STR    __REG(IMX_SSI1_BASE + 0x30)
-+#define SSI1_SOR    __REG(IMX_SSI1_BASE + 0x34)
-+#define SSI1_SACNT  __REG(IMX_SSI1_BASE + 0x38)
-+#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c)
-+#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40)
-+#define SSI1_SATAG  __REG(IMX_SSI1_BASE + 0x44)
-+#define SSI1_STMSK  __REG(IMX_SSI1_BASE + 0x48)
-+#define SSI1_SRMSK  __REG(IMX_SSI1_BASE + 0x4c)
-+
-+#define SSI2_STX0   __REG(IMX_SSI2_BASE + 0x00)
-+#define SSI2_STX0_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x00)
-+#define SSI2_STX1   __REG(IMX_SSI2_BASE + 0x04)
-+#define SSI2_STX1_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x04)
-+#define SSI2_SRX0   __REG(IMX_SSI2_BASE + 0x08)
-+#define SSI2_SRX0_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x08)
-+#define SSI2_SRX1   __REG(IMX_SSI2_BASE + 0x0c)
-+#define SSI2_SRX1_PHYS   __PHYS_REG(IMX_SSI2_BASE + 0x0c)
-+#define SSI2_SCR    __REG(IMX_SSI2_BASE + 0x10)
-+#define SSI2_SISR   __REG(IMX_SSI2_BASE + 0x14)
-+#define SSI2_SIER   __REG(IMX_SSI2_BASE + 0x18)
-+#define SSI2_STCR   __REG(IMX_SSI2_BASE + 0x1c)
-+#define SSI2_SRCR   __REG(IMX_SSI2_BASE + 0x20)
-+#define SSI2_STCCR  __REG(IMX_SSI2_BASE + 0x24)
-+#define SSI2_SRCCR  __REG(IMX_SSI2_BASE + 0x28)
-+#define SSI2_SFCSR  __REG(IMX_SSI2_BASE + 0x2c)
-+#define SSI2_STR    __REG(IMX_SSI2_BASE + 0x30)
-+#define SSI2_SOR    __REG(IMX_SSI2_BASE + 0x34)
-+#define SSI2_SACNT  __REG(IMX_SSI2_BASE + 0x38)
-+#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c)
-+#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40)
-+#define SSI2_SATAG  __REG(IMX_SSI2_BASE + 0x44)
-+#define SSI2_STMSK  __REG(IMX_SSI2_BASE + 0x48)
-+#define SSI2_SRMSK  __REG(IMX_SSI2_BASE + 0x4c)
-+
-+#define SSI_SCR_CLK_IST        (1 << 9)
-+#define SSI_SCR_TCH_EN         (1 << 8)
-+#define SSI_SCR_SYS_CLK_EN     (1 << 7)
-+#define SSI_SCR_I2S_MODE_NORM  (0 << 5)
-+#define SSI_SCR_I2S_MODE_MSTR  (1 << 5)
-+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
-+#define SSI_SCR_SYN            (1 << 4)
-+#define SSI_SCR_NET            (1 << 3)
-+#define SSI_SCR_RE             (1 << 2)
-+#define SSI_SCR_TE             (1 << 1)
-+#define SSI_SCR_SSIEN          (1 << 0)
-+
-+#define SSI_SISR_CMDAU         (1 << 18)
-+#define SSI_SISR_CMDDU         (1 << 17)
-+#define SSI_SISR_RXT           (1 << 16)
-+#define SSI_SISR_RDR1          (1 << 15)
-+#define SSI_SISR_RDR0          (1 << 14)
-+#define SSI_SISR_TDE1          (1 << 13)
-+#define SSI_SISR_TDE0          (1 << 12)
-+#define SSI_SISR_ROE1          (1 << 11)
-+#define SSI_SISR_ROE0          (1 << 10)
-+#define SSI_SISR_TUE1          (1 << 9)
-+#define SSI_SISR_TUE0          (1 << 8)
-+#define SSI_SISR_TFS           (1 << 7)
-+#define SSI_SISR_RFS           (1 << 6)
-+#define SSI_SISR_TLS           (1 << 5)
-+#define SSI_SISR_RLS           (1 << 4)
-+#define SSI_SISR_RFF1          (1 << 3)
-+#define SSI_SISR_RFF0          (1 << 2)
-+#define SSI_SISR_TFE1          (1 << 1)
-+#define SSI_SISR_TFE0          (1 << 0)
-+
-+#define SSI_SIER_RDMAE         (1 << 22)
-+#define SSI_SIER_RIE           (1 << 21)
-+#define SSI_SIER_TDMAE         (1 << 20)
-+#define SSI_SIER_TIE           (1 << 19)
-+#define SSI_SIER_CMDAU_EN      (1 << 18)
-+#define SSI_SIER_CMDDU_EN      (1 << 17)
-+#define SSI_SIER_RXT_EN        (1 << 16)
-+#define SSI_SIER_RDR1_EN       (1 << 15)
-+#define SSI_SIER_RDR0_EN       (1 << 14)
-+#define SSI_SIER_TDE1_EN       (1 << 13)
-+#define SSI_SIER_TDE0_EN       (1 << 12)
-+#define SSI_SIER_ROE1_EN       (1 << 11)
-+#define SSI_SIER_ROE0_EN       (1 << 10)
-+#define SSI_SIER_TUE1_EN       (1 << 9)
-+#define SSI_SIER_TUE0_EN       (1 << 8)
-+#define SSI_SIER_TFS_EN        (1 << 7)
-+#define SSI_SIER_RFS_EN        (1 << 6)
-+#define SSI_SIER_TLS_EN        (1 << 5)
-+#define SSI_SIER_RLS_EN        (1 << 4)
-+#define SSI_SIER_RFF1_EN       (1 << 3)
-+#define SSI_SIER_RFF0_EN       (1 << 2)
-+#define SSI_SIER_TFE1_EN       (1 << 1)
-+#define SSI_SIER_TFE0_EN       (1 << 0)
-+
-+#define SSI_STCR_TXBIT0        (1 << 9)
-+#define SSI_STCR_TFEN1         (1 << 8)
-+#define SSI_STCR_TFEN0         (1 << 7)
-+#define SSI_STCR_TFDIR         (1 << 6)
-+#define SSI_STCR_TXDIR         (1 << 5)
-+#define SSI_STCR_TSHFD         (1 << 4)
-+#define SSI_STCR_TSCKP         (1 << 3)
-+#define SSI_STCR_TFSI          (1 << 2)
-+#define SSI_STCR_TFSL          (1 << 1)
-+#define SSI_STCR_TEFS          (1 << 0)
-+
-+#define SSI_SRCR_RXBIT0        (1 << 9)
-+#define SSI_SRCR_RFEN1         (1 << 8)
-+#define SSI_SRCR_RFEN0         (1 << 7)
-+#define SSI_SRCR_RFDIR         (1 << 6)
-+#define SSI_SRCR_RXDIR         (1 << 5)
-+#define SSI_SRCR_RSHFD         (1 << 4)
-+#define SSI_SRCR_RSCKP         (1 << 3)
-+#define SSI_SRCR_RFSI          (1 << 2)
-+#define SSI_SRCR_RFSL          (1 << 1)
-+#define SSI_SRCR_REFS          (1 << 0)
-+
-+#define SSI_STCCR_DIV2         (1 << 18)
-+#define SSI_STCCR_PSR          (1 << 15)
-+#define SSI_STCCR_WL(x)        ((((x) - 2) >> 1) << 13)
-+#define SSI_STCCR_DC(x)        (((x) & 0x1f) << 8)
-+#define SSI_STCCR_PM(x)        (((x) & 0xff) << 0)
-+
-+#define SSI_SRCCR_DIV2         (1 << 18)
-+#define SSI_SRCCR_PSR          (1 << 15)
-+#define SSI_SRCCR_WL(x)        ((((x) - 2) >> 1) << 13)
-+#define SSI_SRCCR_DC(x)        (((x) & 0x1f) << 8)
-+#define SSI_SRCCR_PM(x)        (((x) & 0xff) << 0)
-+
-+
-+#define SSI_SFCSR_RFCNT1(x)   (((x) & 0xf) << 28)
-+#define SSI_SFCSR_TFCNT1(x)   (((x) & 0xf) << 24)
-+#define SSI_SFCSR_RFWM1(x)    (((x) & 0xf) << 20)
-+#define SSI_SFCSR_TFWM1(x)    (((x) & 0xf) << 16)
-+#define SSI_SFCSR_RFCNT0(x)   (((x) & 0xf) << 12)
-+#define SSI_SFCSR_TFCNT0(x)   (((x) & 0xf) <<  8)
-+#define SSI_SFCSR_RFWM0(x)    (((x) & 0xf) <<  4)
-+#define SSI_SFCSR_TFWM0(x)    (((x) & 0xf) <<  0)
-+
-+#define SSI_STR_TEST          (1 << 15)
-+#define SSI_STR_RCK2TCK       (1 << 14)
-+#define SSI_STR_RFS2TFS       (1 << 13)
-+#define SSI_STR_RXSTATE(x)    (((x) & 0xf) << 8)
-+#define SSI_STR_TXD2RXD       (1 <<  7)
-+#define SSI_STR_TCK2RCK       (1 <<  6)
-+#define SSI_STR_TFS2RFS       (1 <<  5)
-+#define SSI_STR_TXSTATE(x)    (((x) & 0xf) << 0)
-+
-+#define SSI_SOR_CLKOFF        (1 << 6)
-+#define SSI_SOR_RX_CLR        (1 << 5)
-+#define SSI_SOR_TX_CLR        (1 << 4)
-+#define SSI_SOR_INIT          (1 << 3)
-+#define SSI_SOR_WAIT(x)       (((x) & 0x3) << 1)
-+#define SSI_SOR_SYNRST        (1 << 0)
-+
-+#define SSI_SACNT_FRDIV(x)    (((x) & 0x3f) << 5)
-+#define SSI_SACNT_WR          (x << 4)
-+#define SSI_SACNT_RD          (x << 3)
-+#define SSI_SACNT_TIF         (x << 2)
-+#define SSI_SACNT_FV          (x << 1)
-+#define SSI_SACNT_A97EN       (x << 0)
-+
-+
-+/* AUDMUX registers */
-+#define AUDMUX_HPCR1         __REG(IMX_AUDMUX_BASE + 0x00)
-+#define AUDMUX_HPCR2         __REG(IMX_AUDMUX_BASE + 0x04)
-+#define AUDMUX_HPCR3         __REG(IMX_AUDMUX_BASE + 0x08)
-+#define AUDMUX_PPCR1         __REG(IMX_AUDMUX_BASE + 0x10)
-+#define AUDMUX_PPCR2         __REG(IMX_AUDMUX_BASE + 0x14)
-+#define AUDMUX_PPCR3         __REG(IMX_AUDMUX_BASE + 0x18)
-+
-+#define AUDMUX_HPCR_TFSDIR         (1 << 31)
-+#define AUDMUX_HPCR_TCLKDIR        (1 << 30)
-+#define AUDMUX_HPCR_TFCSEL_TX      (0 << 26)
-+#define AUDMUX_HPCR_TFCSEL_RX      (8 << 26)
-+#define AUDMUX_HPCR_TFCSEL(x)      (((x) & 0x7) << 26)
-+#define AUDMUX_HPCR_RFSDIR         (1 << 25)
-+#define AUDMUX_HPCR_RCLKDIR        (1 << 24)
-+#define AUDMUX_HPCR_RFCSEL_TX      (0 << 20)
-+#define AUDMUX_HPCR_RFCSEL_RX      (8 << 20)
-+#define AUDMUX_HPCR_RFCSEL(x)      (((x) & 0x7) << 20)
-+#define AUDMUX_HPCR_RXDSEL(x)      (((x) & 0x7) << 13)
-+#define AUDMUX_HPCR_SYN            (1 << 12)
-+#define AUDMUX_HPCR_TXRXEN         (1 << 10)
-+#define AUDMUX_HPCR_INMEN          (1 <<  8)
-+#define AUDMUX_HPCR_INMMASK(x)     (((x) & 0xff) << 0)
-+
-+#define AUDMUX_PPCR_TFSDIR         (1 << 31)
-+#define AUDMUX_PPCR_TCLKDIR        (1 << 30)
-+#define AUDMUX_PPCR_TFCSEL_TX      (0 << 26)
-+#define AUDMUX_PPCR_TFCSEL_RX      (8 << 26)
-+#define AUDMUX_PPCR_TFCSEL(x)      (((x) & 0x7) << 26)
-+#define AUDMUX_PPCR_RFSDIR         (1 << 25)
-+#define AUDMUX_PPCR_RCLKDIR        (1 << 24)
-+#define AUDMUX_PPCR_RFCSEL_TX      (0 << 20)
-+#define AUDMUX_PPCR_RFCSEL_RX      (8 << 20)
-+#define AUDMUX_PPCR_RFCSEL(x)      (((x) & 0x7) << 20)
-+#define AUDMUX_PPCR_RXDSEL(x)      (((x) & 0x7) << 13)
-+#define AUDMUX_PPCR_SYN            (1 << 12)
-+#define AUDMUX_PPCR_TXRXEN         (1 << 10)
-+
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/imx/imx31-pcm.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx31-pcm.c
-@@ -0,0 +1,417 @@
-+/*
-+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
-+ *
-+ * Copyright 2006 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ *         liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * Based on pxa2xx-pcm.c by	Nicolas Pitre, (C) 2004 MontaVista Software, Inc.
-+ * and on mxc-alsa-mc13783 (C) 2006 Freescale.
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ *
-+ *  Revision history
-+ *    29th Aug 2006   Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/platform_device.h>
-+#include <linux/slab.h>
-+#include <linux/dma-mapping.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <asm/arch/dma.h>
-+#include <asm/arch/spba.h>
-+#include <asm/arch/clock.h>
-+#include <asm/mach-types.h>
-+#include <asm/hardware.h>
-+
-+#include "imx31-pcm.h"
-+
-+/* debug */
-+#define IMX_DEBUG 0
-+#if IMX_DEBUG
-+#define dbg(format, arg...) printk(format, ## arg)
-+#else
-+#define dbg(format, arg...)
-+#endif
-+
-+static const struct snd_pcm_hardware mxc_pcm_hardware = {
-+	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
-+				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-+				   SNDRV_PCM_INFO_MMAP |
-+				   SNDRV_PCM_INFO_MMAP_VALID |
-+				   SNDRV_PCM_INFO_PAUSE |
-+				   SNDRV_PCM_INFO_RESUME),
-+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-+					SNDRV_PCM_FMTBIT_S24_LE,
-+	.buffer_bytes_max	= 32 * 1024,
-+	.period_bytes_min	= 64,
-+	.period_bytes_max	= 8 * 1024,
-+	.periods_min		= 2,
-+	.periods_max		= 255,
-+	.fifo_size		= 0,
-+};
-+
-+struct mxc_runtime_data {
-+	int dma_ch;
-+	struct mxc_pcm_dma_param *dma_params;
-+	spinlock_t dma_lock;
-+	int active, period, periods;
-+	int dma_wchannel;
-+	int tx_spin, rx_spin;
-+	int old_offset;
-+};
-+
-+/*!
-+  * This function stops the current dma transfer for playback
-+  * and clears the dma pointers.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  */
-+static void audio_stop_dma(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	unsigned int offset = dma_size * runtime->periods;
-+	unsigned long flags;
-+
-+	spin_lock_irqsave(&prtd->dma_lock, flags);
-+
-+	dbg("MXC : audio_stop_dma active = 0\n");
-+	prtd->active = 0;
-+	prtd->period = 0;
-+	prtd->periods = 0;
-+
-+	/* this stops the dma channel and clears the buffer ptrs */
-+	mxc_dma_stop(prtd->dma_wchannel);
-+	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_TO_DEVICE);
-+	else
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_FROM_DEVICE);
-+
-+	spin_unlock_irqrestore(&prtd->dma_lock, flags);
-+}
-+
-+/*!
-+  * This function is called whenever a new audio block needs to be
-+  * transferred to mc13783. The function receives the address and the size
-+  * of the new block and start a new DMA transfer.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  */
-+static int dma_new_period(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime =  substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int dma_size;
-+	unsigned int offset;
-+	int ret = 0;
-+	dma_request_t sdma_request;
-+
-+	if (prtd->active){
-+		memset(&sdma_request, 0, sizeof(dma_request_t));
-+		dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	    dbg("s->period (%x) runtime->periods (%d)\n",
-+			s->period,runtime->periods);
-+		dbg("runtime->period_size (%d) dma_size (%d)\n",
-+			(unsigned int)runtime->period_size,
-+			runtime->dma_bytes);
-+
-+       	offset = dma_size * prtd->period;
-+//		snd_assert(dma_size <= DMA_BUF_SIZE, );
-+		if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+			sdma_request.sourceAddr = (char*)(dma_map_single(NULL,
-+				runtime->dma_area + offset, dma_size, DMA_TO_DEVICE));
-+		else
-+			sdma_request.destAddr = (char*)(dma_map_single(NULL,
-+				runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE));
-+		sdma_request.count = dma_size;
-+
-+		dbg("MXC: Start DMA offset (%d) size (%d)\n", offset,
-+						 runtime->dma_bytes);
-+
-+       	mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0);
-+		if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) {
-+			dbg("audio_process_dma: cannot queue DMA buffer\
-+							(%i)\n", ret);
-+			return ret;
-+		}
-+		prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */
-+		prtd->period++;
-+		prtd->period %= runtime->periods;
-+    }
-+	return ret;
-+}
-+
-+
-+/*!
-+  * This is a callback which will be called
-+  * when a TX transfer finishes. The call occurs
-+  * in interrupt context.
-+  *
-+  * @param	dat	pointer to the structure of the current stream.
-+  *
-+  */
-+static void audio_dma_irq(void *data)
-+{
-+	struct snd_pcm_substream *substream;
-+	struct snd_pcm_runtime *runtime;
-+	struct mxc_runtime_data *prtd;
-+	unsigned int dma_size;
-+	unsigned int previous_period;
-+	unsigned int offset;
-+
-+	substream = data;
-+	runtime = substream->runtime;
-+	prtd = runtime->private_data;
-+	previous_period  = prtd->periods;
-+	dma_size = frames_to_bytes(runtime, runtime->period_size);
-+	offset = dma_size * previous_period;
-+
-+	prtd->tx_spin = 0;
-+	prtd->periods++;
-+	prtd->periods %= runtime->periods;
-+
-+	/*
-+	  * Give back to the CPU the access to the non cached memory
-+	  */
-+	if(substream == SNDRV_PCM_STREAM_PLAYBACK)
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_TO_DEVICE);
-+	else
-+		dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size,
-+							DMA_FROM_DEVICE);
-+	/*
-+	  * If we are getting a callback for an active stream then we inform
-+	  * the PCM middle layer we've finished a period
-+	  */
-+ 	if (prtd->active)
-+		snd_pcm_period_elapsed(substream);
-+
-+	/*
-+	  * Trig next DMA transfer
-+	  */
-+	dma_new_period(substream);
-+}
-+
-+/*!
-+  * This function configures the hardware to allow audio
-+  * playback operations. It is called by ALSA framework.
-+  *
-+  * @param	substream	pointer to the structure of the current stream.
-+  *
-+  * @return              0 on success, -1 otherwise.
-+  */
-+static int
-+mxc_pcm_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime =  substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	dma_channel_params *params = rtd->dai->cpu_dai->dma_data;
-+	int ret = 0, channel = 0; // passed in ?;
-+
-+	prtd->period = 0;
-+	prtd->periods = 0;
-+
-+	if(substream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		ret  = mxc_request_dma(&channel, "ALSA TX SDMA");
-+		if (ret < 0) {
-+			dbg("error requesting a write dma channel\n");
-+			return ret;
-+		}
-+
-+	} else {
-+		ret = mxc_request_dma(&channel, "ALSA RX SDMA");
-+		if (ret < 0) {
-+			dbg("error requesting a read dma channel\n");
-+			return ret;
-+		}
-+	}
-+
-+	/* set up chn with params */
-+	params->callback = audio_dma_irq;
-+	mxc_dma_setup_channel(channel, params);
-+	prtd->dma_wchannel = channel;
-+
-+	return ret;
-+}
-+
-+static int mxc_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	int ret;
-+
-+	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
-+	if(ret < 0)
-+		return ret;
-+	runtime->dma_addr = virt_to_phys(runtime->dma_area);
-+
-+	return ret;
-+}
-+
-+static int mxc_pcm_hw_free(struct snd_pcm_substream *substream)
-+{
-+	return snd_pcm_lib_free_pages(substream);
-+}
-+
-+static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct mxc_runtime_data *prtd = substream->runtime->private_data;
-+	int ret = 0;
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+		prtd->tx_spin = 0;
-+		/* requested stream startup */
-+		prtd->active = 1;
-+        ret = dma_new_period(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+		/* requested stream shutdown */
-+		audio_stop_dma(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+		prtd->active = 0;
-+		prtd->periods = 0;
-+		break;
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+		prtd->active = 1;
-+		prtd->tx_spin = 0;
-+		ret = dma_new_period(substream);
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		prtd->active = 0;
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		prtd->active = 1;
-+		if (prtd->old_offset) {
-+			prtd->tx_spin = 0;
-+            ret = dma_new_period(substream);
-+		}
-+		break;
-+	default:
-+		ret = -EINVAL;
-+		break;
-+	}
-+
-+	return ret;
-+}
-+
-+static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+	unsigned int offset = 0;
-+
-+	/* tx_spin value is used here to check if a transfert is active */
-+	if (prtd->tx_spin){
-+		offset = (runtime->period_size * (prtd->periods)) +
-+						(runtime->period_size >> 1);
-+		if (offset >= runtime->buffer_size)
-+			offset = runtime->period_size >> 1;
-+	} else {
-+		offset = (runtime->period_size * (prtd->periods));
-+		if (offset >= runtime->buffer_size)
-+			offset = 0;
-+	}
-+
-+	return offset;
-+}
-+
-+
-+static int mxc_pcm_open(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd;
-+	int ret;
-+
-+	snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware);
-+
-+	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
-+	if (ret < 0)
-+		return ret;
-+	//ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-+	//	&hw_playback_rates);
-+	//if (ret < 0)
-+	//	return ret;
-+
-+	prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL);
-+	if(prtd == NULL)
-+		return -ENOMEM;
-+
-+	runtime->private_data = prtd;
-+	return 0;
-+}
-+
-+static int mxc_pcm_close(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct mxc_runtime_data *prtd = runtime->private_data;
-+
-+	kfree(prtd);
-+	return 0;
-+}
-+
-+static int
-+mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-+				     runtime->dma_area,
-+				     runtime->dma_addr,
-+				     runtime->dma_bytes);
-+}
-+
-+struct snd_pcm_ops mxc_pcm_ops = {
-+	.open		= mxc_pcm_open,
-+	.close		= mxc_pcm_close,
-+	.ioctl		= snd_pcm_lib_ioctl,
-+	.hw_params	= mxc_pcm_hw_params,
-+	.hw_free	= mxc_pcm_hw_free,
-+	.prepare	= mxc_pcm_prepare,
-+	.trigger	= mxc_pcm_trigger,
-+	.pointer	= mxc_pcm_pointer,
-+	.mmap		= mxc_pcm_mmap,
-+};
-+
-+static u64 mxc_pcm_dmamask = 0xffffffff;
-+
-+int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
-+	struct snd_pcm *pcm)
-+{
-+	int ret = 0;
-+
-+	if (!card->dev->dma_mask)
-+		card->dev->dma_mask = &mxc_pcm_dmamask;
-+	if (!card->dev->coherent_dma_mask)
-+		card->dev->coherent_dma_mask = 0xffffffff;
-+
-+	return ret;
-+}
-+
-+struct snd_soc_platform mxc_soc_platform = {
-+	.name		= "mxc-audio",
-+	.pcm_ops 	= &mxc_pcm_ops,
-+	.pcm_new	= mxc_pcm_new,
-+};
-+
-+EXPORT_SYMBOL_GPL(mxc_soc_platform);
-+
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_DESCRIPTION("Freescale i.MX31 PCM DMA module");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/imx/imx31-pcm.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/imx/imx31-pcm.h
-@@ -0,0 +1,241 @@
-+/*
-+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _MXC_PCM_H
-+#define _MXC_PCM_H
-+
-+#include <asm/arch/dma.h>
-+
-+/* temp until imx-regs.h is up2date */
-+#define SSI1_STX0   (SSI1_BASE_ADDR + 0x00)
-+#define SSI1_STX0_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x00)
-+#define SSI1_STX1   (SSI1_BASE_ADDR + 0x04)
-+#define SSI1_STX1_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x04)
-+#define SSI1_SRX0   (SSI1_BASE_ADDR + 0x08)
-+#define SSI1_SRX0_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x08)
-+#define SSI1_SRX1   (SSI1_BASE_ADDR + 0x0c)
-+#define SSI1_SRX1_PHYS   __PHYS_REG(SSI1_BASE_ADDR + 0x0c)
-+#define SSI1_SCR    (SSI1_BASE_ADDR + 0x10)
-+#define SSI1_SISR   (SSI1_BASE_ADDR + 0x14)
-+#define SSI1_SIER   (SSI1_BASE_ADDR + 0x18)
-+#define SSI1_STCR   (SSI1_BASE_ADDR + 0x1c)
-+#define SSI1_SRCR   (SSI1_BASE_ADDR + 0x20)
-+#define SSI1_STCCR  (SSI1_BASE_ADDR + 0x24)
-+#define SSI1_SRCCR  (SSI1_BASE_ADDR + 0x28)
-+#define SSI1_SFCSR  (SSI1_BASE_ADDR + 0x2c)
-+#define SSI1_STR    (SSI1_BASE_ADDR + 0x30)
-+#define SSI1_SOR    (SSI1_BASE_ADDR + 0x34)
-+#define SSI1_SACNT  (SSI1_BASE_ADDR + 0x38)
-+#define SSI1_SACADD (SSI1_BASE_ADDR + 0x3c)
-+#define SSI1_SACDAT (SSI1_BASE_ADDR + 0x40)
-+#define SSI1_SATAG  (SSI1_BASE_ADDR + 0x44)
-+#define SSI1_STMSK  (SSI1_BASE_ADDR + 0x48)
-+#define SSI1_SRMSK  (SSI1_BASE_ADDR + 0x4c)
-+
-+#define SSI2_STX0   (SSI2_BASE_ADDR + 0x00)
-+#define SSI2_STX0_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x00)
-+#define SSI2_STX1   (SSI2_BASE_ADDR + 0x04)
-+#define SSI2_STX1_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x04)
-+#define SSI2_SRX0   (SSI2_BASE_ADDR + 0x08)
-+#define SSI2_SRX0_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x08)
-+#define SSI2_SRX1   (SSI2_BASE_ADDR + 0x0c)
-+#define SSI2_SRX1_PHYS   __PHYS_REG(SSI2_BASE_ADDR + 0x0c)
-+#define SSI2_SCR    (SSI2_BASE_ADDR + 0x10)
-+#define SSI2_SISR   (SSI2_BASE_ADDR + 0x14)
-+#define SSI2_SIER   (SSI2_BASE_ADDR + 0x18)
-+#define SSI2_STCR   (SSI2_BASE_ADDR + 0x1c)
-+#define SSI2_SRCR   (SSI2_BASE_ADDR + 0x20)
-+#define SSI2_STCCR  (SSI2_BASE_ADDR + 0x24)
-+#define SSI2_SRCCR  (SSI2_BASE_ADDR + 0x28)
-+#define SSI2_SFCSR  (SSI2_BASE_ADDR + 0x2c)
-+#define SSI2_STR    (SSI2_BASE_ADDR + 0x30)
-+#define SSI2_SOR    (SSI2_BASE_ADDR + 0x34)
-+#define SSI2_SACNT  (SSI2_BASE_ADDR + 0x38)
-+#define SSI2_SACADD (SSI2_BASE_ADDR + 0x3c)
-+#define SSI2_SACDAT (SSI2_BASE_ADDR + 0x40)
-+#define SSI2_SATAG  (SSI2_BASE_ADDR + 0x44)
-+#define SSI2_STMSK  (SSI2_BASE_ADDR + 0x48)
-+#define SSI2_SRMSK  (SSI2_BASE_ADDR + 0x4c)
-+
-+#define SSI_SCR_CLK_IST        (1 << 9)
-+#define SSI_SCR_TCH_EN         (1 << 8)
-+#define SSI_SCR_SYS_CLK_EN     (1 << 7)
-+#define SSI_SCR_I2S_MODE_NORM  (0 << 5)
-+#define SSI_SCR_I2S_MODE_MSTR  (1 << 5)
-+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5)
-+#define SSI_SCR_SYN            (1 << 4)
-+#define SSI_SCR_NET            (1 << 3)
-+#define SSI_SCR_RE             (1 << 2)
-+#define SSI_SCR_TE             (1 << 1)
-+#define SSI_SCR_SSIEN          (1 << 0)
-+
-+#define SSI_SISR_CMDAU         (1 << 18)
-+#define SSI_SISR_CMDDU         (1 << 17)
-+#define SSI_SISR_RXT           (1 << 16)
-+#define SSI_SISR_RDR1          (1 << 15)
-+#define SSI_SISR_RDR0          (1 << 14)
-+#define SSI_SISR_TDE1          (1 << 13)
-+#define SSI_SISR_TDE0          (1 << 12)
-+#define SSI_SISR_ROE1          (1 << 11)
-+#define SSI_SISR_ROE0          (1 << 10)
-+#define SSI_SISR_TUE1          (1 << 9)
-+#define SSI_SISR_TUE0          (1 << 8)
-+#define SSI_SISR_TFS           (1 << 7)
-+#define SSI_SISR_RFS           (1 << 6)
-+#define SSI_SISR_TLS           (1 << 5)
-+#define SSI_SISR_RLS           (1 << 4)
-+#define SSI_SISR_RFF1          (1 << 3)
-+#define SSI_SISR_RFF0          (1 << 2)
-+#define SSI_SISR_TFE1          (1 << 1)
-+#define SSI_SISR_TFE0          (1 << 0)
-+
-+#define SSI_SIER_RDMAE         (1 << 22)
-+#define SSI_SIER_RIE           (1 << 21)
-+#define SSI_SIER_TDMAE         (1 << 20)
-+#define SSI_SIER_TIE           (1 << 19)
-+#define SSI_SIER_CMDAU_EN      (1 << 18)
-+#define SSI_SIER_CMDDU_EN      (1 << 17)
-+#define SSI_SIER_RXT_EN        (1 << 16)
-+#define SSI_SIER_RDR1_EN       (1 << 15)
-+#define SSI_SIER_RDR0_EN       (1 << 14)
-+#define SSI_SIER_TDE1_EN       (1 << 13)
-+#define SSI_SIER_TDE0_EN       (1 << 12)
-+#define SSI_SIER_ROE1_EN       (1 << 11)
-+#define SSI_SIER_ROE0_EN       (1 << 10)
-+#define SSI_SIER_TUE1_EN       (1 << 9)
-+#define SSI_SIER_TUE0_EN       (1 << 8)
-+#define SSI_SIER_TFS_EN        (1 << 7)
-+#define SSI_SIER_RFS_EN        (1 << 6)
-+#define SSI_SIER_TLS_EN        (1 << 5)
-+#define SSI_SIER_RLS_EN        (1 << 4)
-+#define SSI_SIER_RFF1_EN       (1 << 3)
-+#define SSI_SIER_RFF0_EN       (1 << 2)
-+#define SSI_SIER_TFE1_EN       (1 << 1)
-+#define SSI_SIER_TFE0_EN       (1 << 0)
-+
-+#define SSI_STCR_TXBIT0        (1 << 9)
-+#define SSI_STCR_TFEN1         (1 << 8)
-+#define SSI_STCR_TFEN0         (1 << 7)
-+#define SSI_STCR_TFDIR         (1 << 6)
-+#define SSI_STCR_TXDIR         (1 << 5)
-+#define SSI_STCR_TSHFD         (1 << 4)
-+#define SSI_STCR_TSCKP         (1 << 3)
-+#define SSI_STCR_TFSI          (1 << 2)
-+#define SSI_STCR_TFSL          (1 << 1)
-+#define SSI_STCR_TEFS          (1 << 0)
-+
-+#define SSI_SRCR_RXBIT0        (1 << 9)
-+#define SSI_SRCR_RFEN1         (1 << 8)
-+#define SSI_SRCR_RFEN0         (1 << 7)
-+#define SSI_SRCR_RFDIR         (1 << 6)
-+#define SSI_SRCR_RXDIR         (1 << 5)
-+#define SSI_SRCR_RSHFD         (1 << 4)
-+#define SSI_SRCR_RSCKP         (1 << 3)
-+#define SSI_SRCR_RFSI          (1 << 2)
-+#define SSI_SRCR_RFSL          (1 << 1)
-+#define SSI_SRCR_REFS          (1 << 0)
-+
-+#define SSI_STCCR_DIV2         (1 << 18)
-+#define SSI_STCCR_PSR          (1 << 15)
-+#define SSI_STCCR_WL(x)        ((((x) - 2) >> 1) << 13)
-+#define SSI_STCCR_DC(x)        (((x) & 0x1f) << 8)
-+#define SSI_STCCR_PM(x)        (((x) & 0xff) << 0)
-+
-+#define SSI_SRCCR_DIV2         (1 << 18)
-+#define SSI_SRCCR_PSR          (1 << 15)
-+#define SSI_SRCCR_WL(x)        ((((x) - 2) >> 1) << 13)
-+#define SSI_SRCCR_DC(x)        (((x) & 0x1f) << 8)
-+#define SSI_SRCCR_PM(x)        (((x) & 0xff) << 0)
-+
-+
-+#define SSI_SFCSR_RFCNT1(x)   (((x) & 0xf) << 28)
-+#define SSI_SFCSR_TFCNT1(x)   (((x) & 0xf) << 24)
-+#define SSI_SFCSR_RFWM1(x)    (((x) & 0xf) << 20)
-+#define SSI_SFCSR_TFWM1(x)    (((x) & 0xf) << 16)
-+#define SSI_SFCSR_RFCNT0(x)   (((x) & 0xf) << 12)
-+#define SSI_SFCSR_TFCNT0(x)   (((x) & 0xf) <<  8)
-+#define SSI_SFCSR_RFWM0(x)    (((x) & 0xf) <<  4)
-+#define SSI_SFCSR_TFWM0(x)    (((x) & 0xf) <<  0)
-+
-+#define SSI_STR_TEST          (1 << 15)
-+#define SSI_STR_RCK2TCK       (1 << 14)
-+#define SSI_STR_RFS2TFS       (1 << 13)
-+#define SSI_STR_RXSTATE(x)    (((x) & 0xf) << 8)
-+#define SSI_STR_TXD2RXD       (1 <<  7)
-+#define SSI_STR_TCK2RCK       (1 <<  6)
-+#define SSI_STR_TFS2RFS       (1 <<  5)
-+#define SSI_STR_TXSTATE(x)    (((x) & 0xf) << 0)
-+
-+#define SSI_SOR_CLKOFF        (1 << 6)
-+#define SSI_SOR_RX_CLR        (1 << 5)
-+#define SSI_SOR_TX_CLR        (1 << 4)
-+#define SSI_SOR_INIT          (1 << 3)
-+#define SSI_SOR_WAIT(x)       (((x) & 0x3) << 1)
-+#define SSI_SOR_SYNRST        (1 << 0)
-+
-+#define SSI_SACNT_FRDIV(x)    (((x) & 0x3f) << 5)
-+#define SSI_SACNT_WR          (x << 4)
-+#define SSI_SACNT_RD          (x << 3)
-+#define SSI_SACNT_TIF         (x << 2)
-+#define SSI_SACNT_FV          (x << 1)
-+#define SSI_SACNT_AC97EN      (x << 0)
-+
-+
-+/* AUDMUX registers */
-+#define AUDMUX_HPCR1         (IMX_AUDMUX_BASE + 0x00)
-+#define AUDMUX_HPCR2         (IMX_AUDMUX_BASE + 0x04)
-+#define AUDMUX_HPCR3         (IMX_AUDMUX_BASE + 0x08)
-+#define AUDMUX_PPCR1         (IMX_AUDMUX_BASE + 0x10)
-+#define AUDMUX_PPCR2         (IMX_AUDMUX_BASE + 0x14)
-+#define AUDMUX_PPCR3         (IMX_AUDMUX_BASE + 0x18)
-+
-+#define AUDMUX_HPCR_TFSDIR         (1 << 31)
-+#define AUDMUX_HPCR_TCLKDIR        (1 << 30)
-+#define AUDMUX_HPCR_TFCSEL_TX      (0 << 26)
-+#define AUDMUX_HPCR_TFCSEL_RX      (8 << 26)
-+#define AUDMUX_HPCR_TFCSEL(x)      (((x) & 0x7) << 26)
-+#define AUDMUX_HPCR_RFSDIR         (1 << 25)
-+#define AUDMUX_HPCR_RCLKDIR        (1 << 24)
-+#define AUDMUX_HPCR_RFCSEL_TX      (0 << 20)
-+#define AUDMUX_HPCR_RFCSEL_RX      (8 << 20)
-+#define AUDMUX_HPCR_RFCSEL(x)      (((x) & 0x7) << 20)
-+#define AUDMUX_HPCR_RXDSEL(x)      (((x) & 0x7) << 13)
-+#define AUDMUX_HPCR_SYN            (1 << 12)
-+#define AUDMUX_HPCR_TXRXEN         (1 << 10)
-+#define AUDMUX_HPCR_INMEN          (1 <<  8)
-+#define AUDMUX_HPCR_INMMASK(x)     (((x) & 0xff) << 0)
-+
-+#define AUDMUX_PPCR_TFSDIR         (1 << 31)
-+#define AUDMUX_PPCR_TCLKDIR        (1 << 30)
-+#define AUDMUX_PPCR_TFCSEL_TX      (0 << 26)
-+#define AUDMUX_PPCR_TFCSEL_RX      (8 << 26)
-+#define AUDMUX_PPCR_TFCSEL(x)      (((x) & 0x7) << 26)
-+#define AUDMUX_PPCR_RFSDIR         (1 << 25)
-+#define AUDMUX_PPCR_RCLKDIR        (1 << 24)
-+#define AUDMUX_PPCR_RFCSEL_TX      (0 << 20)
-+#define AUDMUX_PPCR_RFCSEL_RX      (8 << 20)
-+#define AUDMUX_PPCR_RFCSEL(x)      (((x) & 0x7) << 20)
-+#define AUDMUX_PPCR_RXDSEL(x)      (((x) & 0x7) << 13)
-+#define AUDMUX_PPCR_SYN            (1 << 12)
-+#define AUDMUX_PPCR_TXRXEN         (1 << 10)
-+
-+#define SDMA_TXFIFO_WATERMARK				0x4
-+#define SDMA_RXFIFO_WATERMARK				0x6
-+
-+struct mxc_pcm_dma_params {
-+	char *name;			/* stream identifier */
-+	dma_channel_params params;
-+};
-+
-+extern struct snd_soc_cpu_dai mxc_ssi_dai[3];
-+
-+/* platform data */
-+extern struct snd_soc_platform mxc_soc_platform;
-+extern struct snd_ac97_bus_ops mxc_ac97_ops;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/pxa/magician.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/magician.c
-@@ -0,0 +1,563 @@
-+/*
-+ * SoC audio for HTC Magician
-+ *
-+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel at gmail.com>
-+ *
-+ * based on spitz.c,
-+ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
-+ *          Richard Purdie <richard at openedhand.com>
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/timer.h>
-+#include <linux/interrupt.h>
-+#include <linux/platform_device.h>
-+#include <linux/delay.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/hardware/scoop.h>
-+#include <asm/arch/pxa-regs.h>
-+#include <asm/arch/hardware.h>
-+#include <asm/arch/magician.h>
-+#include <asm/arch/magician_cpld.h>
-+#include <asm/mach-types.h>
-+#include "../codecs/uda1380.h"
-+#include "pxa2xx-pcm.h"
-+#include "pxa2xx-i2s.h"
-+#include "pxa2xx-ssp.h"
-+
-+#define MAGICIAN_HP_OFF    0
-+#define MAGICIAN_HEADSET   1
-+#define MAGICIAN_HP        2
-+
-+#define MAGICIAN_SPK_ON    0
-+#define MAGICIAN_SPK_OFF   1
-+
-+#define MAGICIAN_MIC       0
-+#define MAGICIAN_MIC_EXT   1
-+#define MAGICIAN_BT_HS     2
-+
-+/*
-+ * SSP GPIO's
-+ */
-+#define GPIO26_SSP1RX_MD	(26 | GPIO_ALT_FN_1_IN)
-+#define GPIO25_SSP1TX_MD	(25 | GPIO_ALT_FN_2_OUT)
-+#define GPIO23_SSP1CLKS_MD	(23 | GPIO_ALT_FN_2_IN)
-+#define GPIO24_SSP1FRMS_MD	(24 | GPIO_ALT_FN_2_IN)
-+#define GPIO23_SSP1CLKM_MD	(23 | GPIO_ALT_FN_2_OUT)
-+#define GPIO24_SSP1FRMM_MD	(24 | GPIO_ALT_FN_2_OUT)
-+#define GPIO53_SSP1SYSCLK_MD	(53 | GPIO_ALT_FN_2_OUT)
-+
-+static int magician_jack_func = MAGICIAN_HP_OFF;
-+static int magician_spk_func = MAGICIAN_SPK_ON;
-+static int magician_in_sel = MAGICIAN_MIC;
-+
-+extern struct platform_device magician_cpld;
-+
-+static void magician_ext_control(struct snd_soc_codec *codec)
-+{
-+	if (magician_spk_func == MAGICIAN_SPK_ON)
-+		snd_soc_dapm_set_endpoint(codec, "Speaker", 1);
-+	else
-+		snd_soc_dapm_set_endpoint(codec, "Speaker", 0);
-+
-+	/* set up jack connection */
-+	switch (magician_jack_func) {
-+	case MAGICIAN_HP:
-+		/* enable and unmute hp jack, disable mic bias */
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
-+		break;
-+	case MAGICIAN_HEADSET:
-+		/* enable mic jack and bias, mute hp */
-+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
-+		break;
-+	case MAGICIAN_HP_OFF:
-+		/* jack removed, everything off */
-+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-+		break;
-+	}
-+#if 0
-+	/* fixme pH5, can we detect and config the correct Mic type ? */
-+	switch(magician_in_sel) {
-+	case MAGICIAN_IN_MIC:
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
-+		break;
-+	case MAGICIAN_IN_MIC_EXT:
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
-+		break;
-+	case MAGICIAN_IN_BT_HS:
-+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-+		break;
-+	}
-+#endif
-+	snd_soc_dapm_sync_endpoints(codec);
-+}
-+
-+static int magician_startup(snd_pcm_substream_t *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec *codec = rtd->socdev->codec;
-+
-+	/* check the jack status at stream startup */
-+	magician_ext_control(codec);
-+
-+	return 0;
-+}
-+
-+/*
-+ * Magician uses SSP port for playback.
-+ */
-+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
-+				       struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	unsigned int acps, acds, div4;
-+	int ret = 0;
-+
-+	/*
-+	 * Rate = SSPSCLK / (word size(16))
-+	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
-+	 */
-+	switch (params_rate(params)) {
-+	case 8000:
-+		acps = 32842000;
-+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_32;	/* wrong - 32 bits/sample */
-+		div4 = PXA2XX_SSP_CLK_SCDB_4;
-+		break;
-+	case 11025:
-+		acps = 5622000;
-+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_8;	/* 16 bits/sample, 1 slot */
-+		div4 = PXA2XX_SSP_CLK_SCDB_4;
-+		break;
-+	case 22050:
-+		acps = 5622000;
-+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
-+		div4 = PXA2XX_SSP_CLK_SCDB_4;
-+		break;
-+	case 44100:
-+		acps = 11345000;
-+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
-+		div4 = PXA2XX_SSP_CLK_SCDB_4;
-+		break;
-+	case 48000:
-+		acps = 12235000;
-+		acds = PXA2XX_SSP_CLK_AUDIO_DIV_4;
-+		div4 = PXA2XX_SSP_CLK_SCDB_4;
-+		break;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
-+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
-+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set audio clock as clock source */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,
-+			SND_SOC_CLOCK_OUT);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the SSP audio system clock ACDS divider */
-+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
-+			PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the SSP audio system clock SCDB divider4 */
-+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
-+			PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set SSP audio pll clock */
-+	ret = cpu_dai->dai_ops.set_pll(cpu_dai, 0, 0, acps);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+/*
-+ * We have to enable the SSP port early so the UDA1380 can flush
-+ * it's register cache. The UDA1380 can only write it's interpolator and
-+ * decimator registers when the link is running.
-+ */
-+static int magician_playback_prepare(struct snd_pcm_substream *substream)
-+{
-+	/* enable SSP clock - is this needed ? */
-+	SSCR0_P(1) |= SSCR0_SSE;
-+
-+	/* FIXME: ENABLE I2S */
-+	SACR0 |= SACR0_BCKD;
-+	SACR0 |= SACR0_ENB;
-+	pxa_set_cken(CKEN8_I2S, 1);
-+
-+	return 0;
-+}
-+
-+static int magician_playback_hw_free(struct snd_pcm_substream *substream)
-+{
-+	/* FIXME: DISABLE I2S */
-+	SACR0 &= ~SACR0_ENB;
-+	SACR0 &= ~SACR0_BCKD;
-+	pxa_set_cken(CKEN8_I2S, 0);
-+	return 0;
-+}
-+
-+/*
-+ * Magician uses I2S for capture.
-+ */
-+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
-+				      struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	int ret = 0;
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai,
-+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
-+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the I2S system clock as output */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
-+			SND_SOC_CLOCK_OUT);
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+/*
-+ * We have to enable the I2S port early so the UDA1380 can flush
-+ * it's register cache. The UDA1380 can only write it's interpolator and
-+ * decimator registers when the link is running.
-+ */
-+static int magician_capture_prepare(struct snd_pcm_substream *substream)
-+{
-+	SACR0 |= SACR0_ENB;
-+	return 0;
-+}
-+
-+static struct snd_soc_ops magician_capture_ops = {
-+	.startup = magician_startup,
-+	.hw_params = magician_capture_hw_params,
-+	.prepare = magician_capture_prepare,
-+};
-+
-+static struct snd_soc_ops magician_playback_ops = {
-+	.startup = magician_startup,
-+	.hw_params = magician_playback_hw_params,
-+	.prepare = magician_playback_prepare,
-+	.hw_free = magician_playback_hw_free,
-+};
-+
-+static int magician_get_jack(snd_kcontrol_t * kcontrol,
-+			     snd_ctl_elem_value_t * ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = magician_jack_func;
-+	return 0;
-+}
-+
-+static int magician_set_jack(snd_kcontrol_t * kcontrol,
-+			     snd_ctl_elem_value_t * ucontrol)
-+{
-+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-+
-+	if (magician_jack_func == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	magician_jack_func = ucontrol->value.integer.value[0];
-+	magician_ext_control(codec);
-+	return 1;
-+}
-+
-+static int magician_get_spk(snd_kcontrol_t * kcontrol,
-+			    snd_ctl_elem_value_t * ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = magician_spk_func;
-+	return 0;
-+}
-+
-+static int magician_set_spk(snd_kcontrol_t * kcontrol,
-+			    snd_ctl_elem_value_t * ucontrol)
-+{
-+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-+
-+	if (magician_spk_func == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	magician_spk_func = ucontrol->value.integer.value[0];
-+	magician_ext_control(codec);
-+	return 1;
-+}
-+
-+static int magician_get_input(snd_kcontrol_t * kcontrol,
-+			      snd_ctl_elem_value_t * ucontrol)
-+{
-+	ucontrol->value.integer.value[0] = magician_in_sel;
-+	return 0;
-+}
-+
-+static int magician_set_input(snd_kcontrol_t * kcontrol,
-+			      snd_ctl_elem_value_t * ucontrol)
-+{
-+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-+
-+	if (magician_in_sel == ucontrol->value.integer.value[0])
-+		return 0;
-+
-+	magician_in_sel = ucontrol->value.integer.value[0];
-+
-+	switch (magician_in_sel) {
-+	case MAGICIAN_MIC:
-+		magician_egpio_disable(&magician_cpld,
-+				       EGPIO_NR_MAGICIAN_IN_SEL0);
-+		magician_egpio_enable(&magician_cpld,
-+				      EGPIO_NR_MAGICIAN_IN_SEL1);
-+		break;
-+	case MAGICIAN_MIC_EXT:
-+		magician_egpio_disable(&magician_cpld,
-+				       EGPIO_NR_MAGICIAN_IN_SEL0);
-+		magician_egpio_disable(&magician_cpld,
-+				       EGPIO_NR_MAGICIAN_IN_SEL1);
-+	}
-+
-+	return 1;
-+}
-+
-+static int magician_spk_power(struct snd_soc_dapm_widget *w, int event)
-+{
-+	if (SND_SOC_DAPM_EVENT_ON(event))
-+		magician_egpio_enable(&magician_cpld,
-+				      EGPIO_NR_MAGICIAN_SPK_POWER);
-+	else
-+		magician_egpio_disable(&magician_cpld,
-+				       EGPIO_NR_MAGICIAN_SPK_POWER);
-+	return 0;
-+}
-+
-+static int magician_hp_power(struct snd_soc_dapm_widget *w, int event)
-+{
-+	if (SND_SOC_DAPM_EVENT_ON(event))
-+		magician_egpio_enable(&magician_cpld,
-+				      EGPIO_NR_MAGICIAN_EP_POWER);
-+	else
-+		magician_egpio_disable(&magician_cpld,
-+				       EGPIO_NR_MAGICIAN_EP_POWER);
-+	return 0;
-+}
-+
-+static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event)
-+{
-+	if (SND_SOC_DAPM_EVENT_ON(event))
-+		magician_egpio_enable(&magician_cpld,
-+			EGPIO_NR_MAGICIAN_MIC_POWER);
-+	else
-+		magician_egpio_disable(&magician_cpld,
-+			EGPIO_NR_MAGICIAN_MIC_POWER);
-+	return 0;
-+}
-+
-+/* magician machine dapm widgets */
-+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
-+	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
-+	SND_SOC_DAPM_MIC("Mic Jack", magician_mic_bias),
-+	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
-+};
-+
-+/* magician machine audio_map */
-+static const char *audio_map[][3] = {
-+
-+	/* headphone connected to VOUTLHP, VOUTRHP */
-+	{"Headphone Jack", NULL, "VOUTLHP"},
-+	{"Headphone Jack", NULL, "VOUTRHP"},
-+
-+	/* ext speaker connected to VOUTL, VOUTR  */
-+	{"Speaker", NULL, "VOUTL"},
-+	{"Speaker", NULL, "VOUTR"},
-+
-+	/* mic is connected to VINM */
-+	{"VINM", NULL, "Mic Jack"},
-+
-+	/* line is connected to VINL, VINR */
-+	{"VINL", NULL, "Line Jack"},
-+	{"VINR", NULL, "Line Jack"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+static const char *jack_function[] = { "Off", "Headset", "Headphone" };
-+static const char *spk_function[] = { "On", "Off" };
-+static const char *input_select[] = { "Internal Mic", "External Mic" };
-+static const struct soc_enum magician_enum[] = {
-+	SOC_ENUM_SINGLE_EXT(4, jack_function),
-+	SOC_ENUM_SINGLE_EXT(2, spk_function),
-+	SOC_ENUM_SINGLE_EXT(2, input_select),
-+};
-+
-+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
-+	SOC_ENUM_EXT("Jack Function", magician_enum[0], magician_get_jack,
-+			magician_set_jack),
-+	SOC_ENUM_EXT("Speaker Function", magician_enum[1], magician_get_spk,
-+			magician_set_spk),
-+	SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input,
-+			magician_set_input),
-+};
-+
-+/*
-+ * Logic for a uda1380 as connected on a HTC Magician
-+ */
-+static int magician_uda1380_init(struct snd_soc_codec *codec)
-+{
-+	int i, err;
-+
-+	/* NC codec pins */
-+	snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
-+	snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
-+
-+	/* Add magician specific controls */
-+	for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&uda1380_magician_controls[i],
-+				codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	/* Add magician specific widgets */
-+	for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
-+	}
-+
-+	/* Set up magician specific audio path interconnects */
-+	for (i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+				audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+/* magician digital audio interface glue - connects codec <--> CPU */
-+static struct snd_soc_dai_link magician_dai[] = {
-+{
-+	.name = "uda1380",
-+	.stream_name = "UDA1380 Playback",
-+	.cpu_dai = &pxa_ssp_dai[0],
-+	.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
-+	.init = magician_uda1380_init,
-+	.ops = &magician_playback_ops,
-+},
-+{
-+	.name = "uda1380",
-+	.stream_name = "UDA1380 Capture",
-+	.cpu_dai = &pxa_i2s_dai,
-+	.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
-+	.ops = &magician_capture_ops,
-+}
-+};
-+
-+/* magician audio machine driver */
-+static struct snd_soc_machine snd_soc_machine_magician = {
-+	.name = "Magician",
-+	.dai_link = magician_dai,
-+	.num_links = ARRAY_SIZE(magician_dai),
-+};
-+
-+/* magician audio private data */
-+static struct uda1380_setup_data magician_uda1380_setup = {
-+	.i2c_address = 0x18,
-+	.dac_clk = UDA1380_DAC_CLK_WSPLL,
-+};
-+
-+/* magician audio subsystem */
-+static struct snd_soc_device magician_snd_devdata = {
-+	.machine = &snd_soc_machine_magician,
-+	.platform = &pxa2xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_uda1380,
-+	.codec_data = &magician_uda1380_setup,
-+};
-+
-+static struct platform_device *magician_snd_device;
-+
-+static int __init magician_init(void)
-+{
-+	int ret;
-+
-+	if (!machine_is_magician())
-+		return -ENODEV;
-+
-+	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
-+
-+	/* we may need to have the clock running here - pH5 */
-+	magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
-+	udelay(5);
-+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
-+
-+	magician_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!magician_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
-+	magician_snd_devdata.dev = &magician_snd_device->dev;
-+	ret = platform_device_add(magician_snd_device);
-+
-+	if (ret)
-+		platform_device_put(magician_snd_device);
-+
-+	pxa_gpio_mode(GPIO53_SSP1SYSCLK_MD);
-+	pxa_gpio_mode(GPIO26_SSP1RX_MD);
-+	pxa_gpio_mode(GPIO25_SSP1TX_MD);
-+	pxa_gpio_mode(GPIO23_SSP1CLKM_MD);
-+	pxa_gpio_mode(GPIO24_SSP1FRMM_MD);
-+
-+	return ret;
-+}
-+
-+static void __exit magician_exit(void)
-+{
-+	platform_device_unregister(magician_snd_device);
-+
-+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
-+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
-+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
-+	magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
-+}
-+
-+module_init(magician_init);
-+module_exit(magician_exit);
-+
-+MODULE_AUTHOR("Philipp Zabel");
-+MODULE_DESCRIPTION("ALSA SoC Magician");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/templates/template-ac97.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-ac97.c
-@@ -0,0 +1,270 @@
-+/*
-+ * ltemplate-ac97.c -- AC97 support for the xxx chip.
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/platform_device.h>
-+#include <linux/interrupt.h>
-+#include <linux/wait.h>
-+#include <linux/delay.h>
-+#include <linux/mutex.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/ac97_codec.h>
-+#include <sound/initval.h>
-+#include <sound/soc.h>
-+
-+#include <asm/irq.h>
-+#include <asm/hardware.h>
-+
-+#include "template-pcm.h"
-+
-+#define AC97_DIR \
-+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
-+
-+#define AC97_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
-+
-+/* DAI description of AC97 controllers capabilities */
-+static struct snd_soc_dai_mode template_ac97_modes[] = {
-+	{
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = AC97_RATES,
-+		.pcmdir = AC97_DIR,
-+	},
-+};
-+
-+/* AC97 controlller reads codec register */
-+static unsigned short template_ac97_read(struct snd_ac97 *ac97,
-+	unsigned short reg)
-+{
-+}
-+
-+/* AC97 controller writes to codec register */
-+static void template_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
-+	unsigned short val)
-+{
-+}
-+
-+/* AC97 controller asserts a warm reset */
-+static void template_ac97_warm_reset(struct snd_ac97 *ac97)
-+{
-+}
-+
-+/* AC97 controller asserts a cold reset */
-+static void template_ac97_cold_reset(struct snd_ac97 *ac97)
-+{
-+}
-+
-+/* AC97 controller operations */
-+struct snd_ac97_bus_ops soc_ac97_ops = {
-+	.read	= template_ac97_read,
-+	.write	= template_ac97_write,
-+	.warm_reset	= template_ac97_warm_reset,
-+	.reset	= template_ac97_cold_reset,
-+};
-+EXPORT_SYMBOL_GPL(soc_ac97_ops);
-+
-+/* DMA structure describing platform specific AC97 DMA for each logical DAI */
-+static struct template_pcm_dma_params template_ac97_pcm_stereo_out = {
-+	.name			= "AC97 PCM Stereo out",
-+	.dev_addr		= __PREG(PCDR),
-+};
-+
-+static struct template_pcm_dma_params template_ac97_pcm_stereo_in = {
-+	.name			= "AC97 PCM Stereo in",
-+	.dev_addr		= __PREG(PCDR),
-+};
-+
-+static struct template_pcm_dma_params template_ac97_pcm_aux_mono_out = {
-+	.name			= "AC97 Aux PCM (Slot 5) Mono out",
-+	.dev_addr		= __PREG(MODR),
-+};
-+
-+static struct template_pcm_dma_params template_ac97_pcm_aux_mono_in = {
-+	.name			= "AC97 Aux PCM (Slot 5) Mono in",
-+	.dev_addr		= __PREG(MODR),
-+};
-+
-+static struct template_pcm_dma_params template_ac97_pcm_mic_mono_in = {
-+	.name			= "AC97 Mic PCM (Slot 6) Mono in",
-+	.dev_addr		= __PREG(MCDR),
-+};
-+
-+#ifdef CONFIG_PM
-+/* suspend the AC97 controller */
-+static int template_ac97_suspend(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+/* resume the AC97 controller */
-+static int template_ac97_resume(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+#else
-+#define template_ac97_suspend	NULL
-+#define template_ac97_resume	NULL
-+#endif
-+
-+/*
-+ * Probe initialises the AC97 controller. e.g.
-+ * request any IRQ's
-+ * configure GPIO's
-+ * enable any clocks
-+ */
-+static int template_ac97_probe(struct platform_device *pdev)
-+{
-+}
-+
-+/*
-+ * Free's resources setup in probe()
-+ */
-+static void template_ac97_remove(struct platform_device *pdev)
-+{
-+}
-+
-+/*
-+ * Alsa operations
-+ * Only implement the required operations for your platform.
-+ * These operations are specific to the AC97 controller and DAI only.
-+ */
-+
-+ /*
-+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
-+ */
-+static int template_ac97_startup(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when a PCM substream is closed. Private data can be
-+ * freed here.
-+ */
-+static int template_ac97_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the PCM substream is prepared, can set format, sample
-+ * rate, etc.  This function is non atomic and can be called multiple times,
-+ * it can refer to the runtime info.
-+ */
-+static int template_ac97_prepare(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the hardware params are set by application. This
-+ * function can also be called multiple times and can allocate buffers
-+ * (using snd_pcm_lib_* ). It's non-atomic.
-+ */
-+static int template_ac97_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+}
-+
-+/*
-+ * Free's resources allocated by hw_params, can be called multiple times
-+ */
-+static int template_ac97_hw_free(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Starts (Triggers) audio playback or capture.
-+ * Usually only needed for DMA
-+ */
-+static int template_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+}
-+
-+/*
-+ * Define each AC97 slot grouping as a DAI.
-+ *
-+ * e.g. (below)
-+ *  Slots 3 & 4 = HiFi DAI
-+ *  Slot 5 = Aux playback
-+ *  Slot 7 = Mic Capture
-+ *
-+ * This gives 3 logical DAI's on the 1 physical AC97 DAI.
-+ *
-+ */
-+struct snd_soc_cpu_dai template_ac97_dai[] = {
-+{
-+	.name = "template-ac97-HiFi",
-+	.id = 0,
-+	.type = SND_SOC_DAI_AC97,
-+	/* DAI driver operations - only needed on 1st logical DAI in AC97 */
-+	.probe = template_ac97_probe,
-+	.remove = template_ac97_remove,
-+	.suspend = template_ac97_suspend,
-+	.resume = template_ac97_resume,
-+	/* playback and capture stream info */
-+	.playback = {
-+		.stream_name = "AC97 Playback",
-+		.channels_min = 2,
-+		.channels_max = 2,},
-+	.capture = {
-+		.stream_name = "AC97 Capture",
-+		.channels_min = 2,
-+		.channels_max = 2,},
-+	/* alsa PCM operations */
-+	.ops = {
-+		.startup = template_ac97_startup,
-+		.shutdown = template_ac97_shutdown,
-+		.prepare = template_ac97_prepare,
-+		.trigger = template_ac97_trigger,
-+		.hw_params = template_ac97_hw_params,
-+		.hw_free = template_ac97_hw_free,},
-+	/* DAI capabilities */
-+	.caps = {
-+		.num_modes = ARRAY_SIZE(template_ac97_modes),
-+		.mode = template_ac97_modes,},
-+},
-+/* AC97 AUX playback - not supported on all controllers */
-+{
-+	.name = "template-ac97-aux",
-+	.id = 1,
-+	.type = SND_SOC_DAI_AC97,
-+	.playback = {
-+		.stream_name = "AC97 Aux Playback",
-+		.channels_min = 1,
-+		.channels_max = 1,},
-+	.capture = {
-+		.stream_name = "AC97 Aux Capture",
-+		.channels_min = 1,
-+		.channels_max = 1,},
-+	.ops = {
-+		.hw_params = template_ac97_hw_aux_params,},
-+	.caps = {
-+		.num_modes = ARRAY_SIZE(template_ac97_modes),
-+		.mode = template_ac97_modes,},
-+},
-+/* AC97 Mic capture - not supported on all controllers */
-+{
-+	.name = "template-ac97-mic",
-+	.id = 2,
-+	.type = SND_SOC_DAI_AC97,
-+	.capture = {
-+		.stream_name = "AC97 Mic Capture",
-+		.channels_min = 1,
-+		.channels_max = 1,},
-+	.ops = {
-+		.hw_params = template_ac97_hw_mic_params,},
-+	.caps = {
-+		.num_modes = ARRAY_SIZE(template_ac97_modes),
-+		.mode = template_ac97_modes,},},
-+};
-+EXPORT_SYMBOL_GPL(template_ac97_dai);
-+
-Index: linux-2.6.21-moko/sound/soc/templates/template-codec.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-codec.c
-@@ -0,0 +1,784 @@
-+/*
-+ * template-codec.c  --  Template Codec  Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "template-codec.h"
-+
-+#define AUDIO_NAME "template-codec"
-+#define TEMPLATE_VERSION "0.1"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define TEMPLATE_DEBUG 0
-+
-+#ifdef TEMPLATE_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_template_codec;
-+
-+/*
-+ * template_codec register cache
-+ */
-+static const u16 template_codec_reg[TEMPLATE_CACHEREGNUM] = {
-+    0x0097, 0x0097, 0x0079, 0x0079,
-+    0x000a, 0x0008, 0x009f, 0x000a,
-+    0x0000, 0x0000
-+};
-+
-+/* Codec DAI can support these hardware formats */
-+#define TEMPLATE_DAIFMT \
-+	(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \
-+	SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \
-+	SND_SOC_DAIFMT_IB_IF)
-+
-+/* Codec DAI supports direction */
-+#define TEMPLATE_DIR \
-+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
-+
-+/* Codec DAI supports rates */
-+#define TEMPLATE_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+/* Codec DAI supports PCM word sizes */
-+#define TEMPLATE_HIFI_BITS \
-+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
-+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-+
-+/*
-+ * Description of supported codec DAI supported modes.
-+ */
-+static struct snd_soc_dai_mode template_codec_modes[] = {
-+	/* codec frame and clock master modes */
-+	/* 8k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 1536,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 2304,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 1408,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 2112,
-+		.bfs = 64,
-+	},
-+
-+	/* 32k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_32000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 384,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_32000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 576,
-+		.bfs = 64,
-+	},
-+
-+	/* 44.1k & 48k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 256,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 384,
-+		.bfs = 64,
-+	},
-+
-+	/* 88.2 & 96k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 128,
-+		.bfs = 64,
-+	},
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.fs = 192,
-+		.bfs = 64,
-+	},
-+
-+	/* USB codec frame and clock master modes */
-+	/* 8k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 1500,
-+		.bfs = SND_SOC_FSBD(1),
-+	},
-+
-+	/* 44.1k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_44100,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 272,
-+		.bfs = SND_SOC_FSBD(1),
-+	},
-+
-+	/* 48k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_48000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 250,
-+		.bfs = SND_SOC_FSBD(1),
-+	},
-+
-+	/* 88.2k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_88200,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 136,
-+		.bfs = SND_SOC_FSBD(1),
-+	},
-+
-+	/* 96k */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = SNDRV_PCM_RATE_96000,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 125,
-+		.bfs = SND_SOC_FSBD(1),
-+	},
-+
-+	/* codec frame and clock slave modes */
-+	{
-+		.fmt = TEMPLATE_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = TEMPLATE_HIFI_BITS,
-+		.pcmrate = TEMPLATE_RATES,
-+		.pcmdir = TEMPLATE_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = SND_SOC_FS_ALL,
-+		.bfs = SND_SOC_FSB_ALL,
-+	},
-+};
-+
-+/*
-+ * read template_codec register cache
-+ */
-+static inline unsigned int template_codec_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+
-+	if (reg >= TEMPLATE_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write template_codec register cache
-+ */
-+static inline void template_codec_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= TEMPLATE_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the template codec register space
-+ */
-+static int template_codec_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* format the data - codec specific */
-+	data[0] = reg;
-+	data[1] = value;
-+
-+	template_codec_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define template_codec_reset(c)	template_codec_write(c, TEMPLATE_RESET, 0)
-+
-+
-+/* template codec non DAPM controls */
-+static const struct snd_kcontrol_new template_codec_snd_controls[] = {
-+};
-+
-+/* add non dapm controls */
-+static int template_codec_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(template_codec_snd_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&template_codec_snd_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* template codec DAPM controls */
-+static const struct snd_soc_dapm_widget template_codec_dapm_widgets[] = {
-+};
-+
-+/*
-+ * template codec audio interconnectiosn between sink and source.
-+ */
-+static const char *audio_map[][3] = {
-+
-+
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int template_codec_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(template_codec_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &template_codec_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+/*
-+ * Configures the codec SYSCLK/MCLK (system or master clock)
-+ */
-+static unsigned int template_codec_config_sysclk(struct snd_soc_codec_dai *dai,
-+	struct snd_soc_clock_info *info, unsigned int clk)
-+{
-+	dai->mclk = 0;
-+
-+	/* check that the calculated FS and rate actually match a clock from
-+	 * the machine driver */
-+	if (info->fs * info->rate == clk)
-+		dai->mclk = clk;
-+
-+	return dai->mclk;
-+}
-+
-+/*
-+ * Alsa operations
-+ * Only implement the required operations for your platform.
-+ * These operations are specific to the codec only.
-+ */
-+
-+ /*
-+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
-+ */
-+static int template_codec_startup(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when a PCM substream is closed. Private data can be
-+ * freed here.
-+ */
-+static int template_codec_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the hardware params are set by application. This
-+ * function can also be called multiple times and can allocate buffers
-+ * (using snd_pcm_lib_* ). It's non-atomic.
-+ */
-+static int template_codec_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+}
-+
-+/*
-+ * Free's resources allocated by hw_params, can be called multiple times
-+ */
-+static int template_codec_hw_free(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Starts (Triggers) audio playback or capture.
-+ * Usually only needed for DMA
-+ */
-+static int template_codec_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the PCM substream is prepared, can set format, sample
-+ * rate, etc.  This function is non atomic and can be called multiple times,
-+ * it can refer to the runtime info.
-+ */
-+static int template_codec_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	/* set master/slave audio interface */
-+	switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	}
-+
-+	/* interface format */
-+	switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		break;
-+	}
-+
-+	/* bit size */
-+	switch (rtd->codec_dai->dai_runtime.pcmfmt) {
-+	case SNDRV_PCM_FMTBIT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FMTBIT_S20_3LE:
-+		break;
-+	case SNDRV_PCM_FMTBIT_S24_LE:
-+		break;
-+	case SNDRV_PCM_FMTBIT_S32_LE:
-+		break;
-+	}
-+
-+	/* clock inversion */
-+	switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		break;
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * Enable / Disable codec digital soft mute
-+ */
-+static int template_codec_mute(struct snd_soc_codec *codec,
-+	struct snd_soc_codec_dai *dai, int mute)
-+{
-+}
-+
-+/*
-+ * Codec DAPM event handler
-+ * This handles codec level DAPM events
-+ */
-+static int template_codec_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+	u16 reg = template_codec_read_reg_cache(codec, TEMPLATE_PWR) & 0xff7f;
-+
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* e.g. vref/mid, osc on, */
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, */
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		break;
-+	}
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+/*
-+ * Define codec DAI.
-+ */
-+struct snd_soc_codec_dai template_codec_dai = {
-+	.name = "codec xxx",
-+	/* playback and capture stream info */
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+	},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+	},
-+	/* codec operations */
-+	.config_sysclk = template_codec_config_sysclk,
-+	.digital_mute = template_codec_mute,
-+	/* alsa PCM operations */
-+	.ops = {
-+		.startup = template_codec_startup,
-+		.shutdown = template_codec_shutdown,
-+		.prepare = template_codec_prepare,
-+		.trigger = template_codec_trigger,
-+		.hw_params = template_codec_hw_params,
-+		.hw_free = template_codec_hw_free,},
-+	/* codec capabilities */
-+	.caps = {
-+		.num_modes = ARRAY_SIZE(template_codec_modes),
-+		.mode = template_codec_modes,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(template_codec_dai);
-+
-+static int template_codec_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	template_codec_write(codec, TEMPLATE_ACTIVE, 0x0);
-+	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int template_codec_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(template_codec_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	template_codec_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the TEMPLATE driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int template_codec_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "TEMPLATE";
-+	codec->owner = THIS_MODULE;
-+	codec->read = template_codec_read_reg_cache;
-+	codec->write = template_codec_write;
-+	codec->dapm_event = template_codec_dapm_event;
-+	codec->dai = &template_codec_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(template_codec_reg);
-+
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(template_codec_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache,
-+		template_codec_reg, sizeof(u16) * ARRAY_SIZE(template_codec_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(template_codec_reg);
-+
-+	template_codec_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		kfree(codec->reg_cache);
-+		return ret;
-+	}
-+
-+	/* power on device */
-+	template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/* set the update bits */
-+	reg = template_codec_read_reg_cache(codec, TEMPLATE_LOUT1V);
-+	template_codec_write(codec, TEMPLATE_LOUT1V, reg | 0x0100);
-+	reg = template_codec_read_reg_cache(codec, TEMPLATE_ROUT1V);
-+	template_codec_write(codec, TEMPLATE_ROUT1V, reg | 0x0100);
-+	reg = template_codec_read_reg_cache(codec, TEMPLATE_LINVOL);
-+	template_codec_write(codec, TEMPLATE_LINVOL, reg | 0x0100);
-+	reg = template_codec_read_reg_cache(codec, TEMPLATE_RINVOL);
-+	template_codec_write(codec, TEMPLATE_RINVOL, reg | 0x0100);
-+
-+	template_codec_add_controls(codec);
-+	template_codec_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+		snd_soc_free_pcms(socdev);
-+		snd_soc_dapm_free(socdev);
-+	}
-+
-+	return ret;
-+}
-+
-+static struct snd_soc_device *template_codec_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * TEMPLATE 2 wire address is determined by GPIO5
-+ * state during powerup.
-+ *    low  = 0x1a
-+ *    high = 0x1b
-+ */
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver template_codec_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int template_codec_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = template_codec_socdev;
-+	struct template_codec_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = template_codec_init(socdev);
-+	if (ret < 0) {
-+		err("failed to initialise TEMPLATE\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int template_codec_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int template_codec_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, template_codec_codec_probe);
-+}
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver template_codec_i2c_driver = {
-+	.driver = {
-+		.name = "TEMPLATE I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_TEMPLATE,
-+	.attach_adapter = template_codec_i2c_attach,
-+	.detach_client =  template_codec_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "TEMPLATE",
-+	.driver = &template_codec_i2c_driver,
-+};
-+#endif
-+
-+static int template_codec_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct template_codec_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("TEMPLATE Audio Codec %s", TEMPLATE_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	template_codec_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&template_codec_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip and remove */
-+static int template_codec_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		template_codec_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&template_codec_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+/* codec device ops */
-+struct snd_soc_codec_device soc_codec_dev_template_codec = {
-+	.probe = 	template_codec_probe,
-+	.remove = 	template_codec_remove,
-+	.suspend = 	template_codec_suspend,
-+	.resume =	template_codec_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_template_codec);
-+
-Index: linux-2.6.21-moko/sound/soc/templates/template-i2s.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-i2s.c
-@@ -0,0 +1,223 @@
-+/*
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/device.h>
-+#include <linux/delay.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/initval.h>
-+#include <sound/soc.h>
-+
-+#include <asm/hardware.h>
-+
-+#include "template-pcm.h"
-+
-+/* supported I2S DAI hardware formats */
-+#define TEMPLATE_I2S_DAIFMT \
-+	(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF)
-+
-+/* supported I2S direction */
-+#define TEMPLATE_I2S_DIR \
-+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
-+
-+/* supported I2S rates */
-+#define TEMPLATE_I2S_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+/* I2S controller DAI capabilities */
-+static struct snd_soc_dai_mode template_i2s_modes[] = {
-+	/* template I2S frame and clock master modes */
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_8000,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_11025,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_16000,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_22050,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_44100,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = SNDRV_PCM_RATE_48000,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.flags = SND_SOC_DAI_BFS_DIV,
-+		.fs = 256,
-+		.bfs = SND_SOC_FSBD(4),
-+	},
-+
-+	/* template I2S frame master and clock slave mode */
-+	{
-+		.fmt = TEMPLATE_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS,
-+		.pcmfmt = SNDRV_PCM_FMTBIT_S16_LE,
-+		.pcmrate = TEMPLATE_I2S_RATES,
-+		.pcmdir = TEMPLATE_I2S_DIR,
-+		.fs = SND_SOC_FS_ALL,
-+		.flags = SND_SOC_DAI_BFS_RATE,
-+		.bfs = 64,
-+	},
-+};
-+
-+/* I2S controller platform specific DMA parameters */
-+static struct template_pcm_dma_params template_i2s_pcm_stereo_out = {
-+	.name			= "I2S PCM Stereo out",
-+	.dev_addr		= __PREG(SADR),
-+};
-+
-+static struct template_pcm_dma_params template_i2s_pcm_stereo_in = {
-+	.name			= "I2S PCM Stereo in",
-+	.dev_addr		= __PREG(SADR),
-+};
-+
-+#ifdef CONFIG_PM
-+/* suspend I2S controller */
-+static int template_i2s_suspend(struct platform_device *dev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+/* resume I2S controller */
-+static int template_i2s_resume(struct platform_device *pdev,
-+	struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+#else
-+#define template_i2s_suspend	NULL
-+#define template_i2s_resume	NULL
-+#endif
-+
-+/* configure the I2S controllers MCLK or SYSCLK */
-+static unsigned int template_i2s_config_sysclk(struct snd_soc_cpu_dai *iface,
-+	struct snd_soc_clock_info *info, unsigned int clk)
-+{
-+}
-+
-+
-+/*
-+ * Alsa operations
-+ * Only implement the required operations for your platform.
-+ * These operations are specific to the I2S controller and DAI only.
-+ */
-+
-+ /*
-+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
-+ */
-+static int template_i2s_startup(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when a PCM substream is closed. Private data can be
-+ * freed here.
-+ */
-+static int template_i2s_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the PCM substream is prepared, can set format, sample
-+ * rate, etc.  This function is non atomic and can be called multiple times,
-+ * it can refer to the runtime info.
-+ */
-+static int template_i2s_prepare(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the hardware params are set by application. This
-+ * function can also be called multiple times and can allocate buffers
-+ * (using snd_pcm_lib_* ). It's non-atomic.
-+ */
-+static int template_i2s_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+}
-+
-+/*
-+ * Free's resources allocated by hw_params, can be called multiple times
-+ */
-+static int template_i2s_hw_free(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Starts (Triggers) audio playback or capture.
-+ * Usually only needed for DMA
-+ */
-+static int template_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+}
-+
-+
-+struct snd_soc_cpu_dai template_i2s_dai = {
-+	.name = "template-i2s",
-+	.id = 0,
-+	.type = SND_SOC_DAI_I2S,
-+	.suspend = template_i2s_suspend,
-+	.resume = template_i2s_resume,
-+	.config_sysclk = template_i2s_config_sysclk,
-+	.playback = {
-+		.channels_min = 2,
-+		.channels_max = 2,},
-+	.capture = {
-+		.channels_min = 2,
-+		.channels_max = 2,},
-+	.ops = {
-+		.startup = template_i2s_startup,
-+		.shutdown = template_i2s_shutdown,
-+		.prepare = template_i2s_prepare,
-+		.trigger = template_i2s_trigger,
-+		.hw_params = template_i2s_hw_params,
-+		.hw_free = template_i2s_hw_free,},
-+	.caps = {
-+		.num_modes = ARRAY_SIZE(template_i2s_modes),
-+		.mode = template_i2s_modes,},
-+};
-+
-+EXPORT_SYMBOL_GPL(template_i2s_dai);
-Index: linux-2.6.21-moko/sound/soc/templates/template-pcm.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-pcm.c
-@@ -0,0 +1,166 @@
-+/*
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/platform_device.h>
-+#include <linux/slab.h>
-+#include <linux/dma-mapping.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+
-+#include <asm/dma.h>
-+#include <asm/hardware.h>
-+
-+#include "template-pcm.h"
-+
-+/* PCM hardware DMA capabilities - platform specific */
-+static const struct snd_pcm_hardware template_pcm_hardware = {
-+	.info			= SNDRV_PCM_INFO_MMAP |
-+				  SNDRV_PCM_INFO_MMAP_VALID |
-+				  SNDRV_PCM_INFO_INTERLEAVED |
-+				  SNDRV_PCM_INFO_PAUSE |
-+				  SNDRV_PCM_INFO_RESUME,
-+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-+					SNDRV_PCM_FMTBIT_S24_LE |
-+					SNDRV_PCM_FMTBIT_S32_LE,
-+	.period_bytes_min	= 32,
-+	.period_bytes_max	= 8192 - 32,
-+	.periods_min		= 1,
-+	.periods_max		= PAGE_SIZE/sizeof(pxa_dma_desc),
-+	.buffer_bytes_max	= 128 * 1024,
-+	.fifo_size		= 32,
-+};
-+
-+/*
-+ * Called by ALSA when the hardware params are set by application. This
-+ * function can also be called multiple times and can allocate buffers
-+ * (using snd_pcm_lib_* ). It's non-atomic.
-+ */
-+static int template_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+}
-+
-+/*
-+ * Free's resources allocated by hw_params, can be called multiple times
-+ */
-+static int template_pcm_hw_free(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the PCM substream is prepared, can set format, sample
-+ * rate, etc.  This function is non atomic and can be called multiple times,
-+ * it can refer to the runtime info.
-+ */
-+static int template_pcm_prepare(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Starts (Triggers) audio playback or capture.
-+ * Usually only needed for DMA
-+ */
-+static int template_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct template_runtime_data *prtd = substream->runtime->private_data;
-+	int ret = 0;
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+		break;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+		break;
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		break;
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+		break;
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		break;
-+	default:
-+		ret = -EINVAL;
-+	}
-+
-+	return ret;
-+}
-+
-+/*
-+ * Returns the DMA audio frame position
-+ */
-+static snd_pcm_uframes_t
-+template_pcm_pointer(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+ /*
-+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
-+ */
-+static int template_pcm_open(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when a PCM substream is closed. Private data can be
-+ * freed here.
-+ */
-+static int template_pcm_close(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/* map DMA audio buffer into user space */
-+static int template_pcm_mmap(struct snd_pcm_substream *substream,
-+	struct vm_area_struct *vma)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-+				     runtime->dma_area,
-+				     runtime->dma_addr,
-+				     runtime->dma_bytes);
-+}
-+
-+/* ALSA PCM operations */
-+struct snd_pcm_ops template_pcm_ops = {
-+	.open		= template_pcm_open,
-+	.close		= template_pcm_close,
-+	.ioctl		= snd_pcm_lib_ioctl,
-+	.hw_params	= template_pcm_hw_params,
-+	.hw_free	= template_pcm_hw_free,
-+	.prepare	= template_pcm_prepare,
-+	.trigger	= template_pcm_trigger,
-+	.pointer	= template_pcm_pointer,
-+	.mmap		= template_pcm_mmap,
-+};
-+
-+/*
-+ * Called by ASoC core to free platform DMA.
-+ */
-+static void template_pcm_free_dma_buffers(struct snd_pcm *pcm)
-+{
-+}
-+
-+/*
-+ * Called by the ASoC core to create and initialise the platform DMA.
-+ */
-+int template_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
-+	struct snd_pcm *pcm)
-+{
-+}
-+
-+/* template audio platform */
-+struct snd_soc_platform template_soc_platform = {
-+	.name		= "template-audio",
-+	.pcm_ops 	= &template_pcm_ops,
-+	.pcm_new	= template_pcm_new,
-+	.pcm_free	= template_pcm_free_dma_buffers,
-+};
-+EXPORT_SYMBOL_GPL(template_soc_platform);
-Index: linux-2.6.21-moko/sound/soc/templates/template-pcm.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-pcm.h
-@@ -0,0 +1,19 @@
-+/*
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _TEMPLATE_PCM_H
-+#define _TEMPLATE_PCM_H
-+
-+/* platform specific structs can be declared here */
-+
-+extern struct snd_soc_cpu_dai template_ac97_dai[3];
-+extern struct snd_soc_cpu_dai template_i2s_dai;
-+
-+/* template platform data */
-+extern struct snd_soc_platform template_soc_platform;
-+extern struct snd_ac97_bus_ops tempalte_ac97_ops;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/templates/template-codec.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-codec.h
-@@ -0,0 +1,21 @@
-+/*
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _TEMPLATE_H
-+#define _TEMPLATE_H
-+
-+/* TEMPLATE register space */
-+
-+#define TEMPLATE_CACHEREGNUM 	10
-+
-+struct template_codec_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai template_codec_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_template_codec;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/templates/template-machine.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/templates/template-machine.c
-@@ -0,0 +1,161 @@
-+/*
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/timer.h>
-+#include <linux/interrupt.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/mach-types.h>
-+
-+#include "../codecs/template-codec.h"
-+#include "template-pcm.h"
-+
-+/*
-+ * Alsa operations
-+ * Only implement the required operations for your platform.
-+ * These operations are specific to the machine only.
-+ */
-+
-+ /*
-+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
-+ */
-+static int template_machine_startup(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when a PCM substream is closed. Private data can be
-+ * freed here.
-+ */
-+static int template_machine_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/*
-+ * Called by ALSA when the hardware params are set by application. This
-+ * function can also be called multiple times and can allocate buffers
-+ * (using snd_pcm_lib_* ). It's non-atomic.
-+ */
-+static int template_machine_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+}
-+
-+/*
-+ * Free's resources allocated by hw_params, can be called multiple times
-+ */
-+static int template_machine_hw_free(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/* machine Alsa PCM operations */
-+static struct snd_soc_ops template_ops = {
-+	.startup = template_machine_startup,
-+	.shutdown = template_machine_shutdown,
-+	.hw_free = template_machine_hw_free,
-+	.hw_params = template_machine_hw_params,
-+};
-+
-+/* machine audio map (connections to the codec pins) */
-+static const char *audio_map[][3] = {
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+/*
-+ * Initialise the machine audio subsystem.
-+ */
-+static int template_machine_init(struct snd_soc_codec *codec)
-+{
-+	/* mark unused codec pins as NC */
-+
-+	/* Add template specific controls */
-+
-+	/* Add template specific widgets */
-+
-+	/* Set up template specific audio path audio_map */
-+
-+	/* synchronise subsystem */
-+	snd_soc_dapm_sync_endpoints(codec);
-+	return 0;
-+}
-+
-+/*
-+ * Configure the clocking within the audio subsystem
-+ */
-+static unsigned int template_config_sysclk(struct snd_soc_pcm_runtime *rtd,
-+	struct snd_soc_clock_info *info)
-+{
-+}
-+
-+/* template digital audio interface glue - connects codec <--> CPU */
-+static struct snd_soc_dai_link template_dai = {
-+	.name = "Codec",
-+	.stream_name = "Stream Name",
-+	.cpu_dai = &template_i2s_dai,
-+	.codec_dai = &template_codec_dai,
-+	.init = template_machine_init,
-+	.config_sysclk = template_config_sysclk,
-+};
-+
-+/* template audio machine driver */
-+static struct snd_soc_machine snd_soc_machine_template = {
-+	.name = "Machine",
-+	.dai_link = &template_dai,
-+	.num_links = ARRAY_SIZE(template_dai),
-+	.ops = &template_ops,
-+};
-+
-+/* template audio private data */
-+static struct codec_priv_setup_data template_codec_setup = {
-+	.i2c_address = 0x1b,
-+};
-+
-+/* template audio subsystem */
-+static struct snd_soc_device template_snd_devdata = {
-+	.machine = &snd_soc_machine_template,
-+	.platform = &template_soc_platform,
-+	.codec_dev = &soc_codec_dev_wm8731,
-+	.codec_data = &template_codec_setup,
-+};
-+
-+static struct platform_device *template_snd_device;
-+
-+static int __init template_init(void)
-+{
-+	int ret;
-+
-+	template_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!template_snd_device)
-+		return -ENOMEM;
-+
-+	platform_set_drvdata(template_snd_device, &template_snd_devdata);
-+	template_snd_devdata.dev = &template_snd_device->dev;
-+	ret = platform_device_add(template_snd_device);
-+
-+	if (ret)
-+		platform_device_put(template_snd_device);
-+
-+	return ret;
-+}
-+
-+static void __exit template_exit(void)
-+{
-+	platform_device_unregister(template_snd_device);
-+}
-+
-+module_init(template_init);
-+module_exit(template_exit);
-+
-Index: linux-2.6.21-moko/sound/soc/pxa/pxa2xx-ssp.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/pxa/pxa2xx-ssp.h
-@@ -0,0 +1,43 @@
-+/*
-+ * linux/sound/arm/pxa2xx-ssp.h
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _PXA2XX_SSP_H
-+#define _PXA2XX_SSP_H
-+
-+/* pxa2xx DAI SSP ID's */
-+#define PXA2XX_DAI_SSP1			0
-+#define PXA2XX_DAI_SSP2			1
-+#define PXA2XX_DAI_SSP3			2
-+
-+/* SSP clock sources */
-+#define PXA2XX_SSP_CLK_PLL	0
-+#define PXA2XX_SSP_CLK_EXT	1
-+#define PXA2XX_SSP_CLK_NET	2
-+#define PXA2XX_SSP_CLK_AUDIO	3
-+#define PXA2XX_SSP_CLK_NET_PLL	4
-+
-+/* SSP audio dividers */
-+#define PXA2XX_SSP_AUDIO_DIV_ACDS		0
-+#define PXA2XX_SSP_AUDIO_DIV_SCDB		1
-+#define PXA2XX_SSP_DIV_SCR				2
-+
-+/* SSP ACDS audio dividers values */
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_1		0
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_2		1
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_4		2
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_8		3
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_16	4
-+#define PXA2XX_SSP_CLK_AUDIO_DIV_32	5
-+
-+/* SSP divider bypass */
-+#define PXA2XX_SSP_CLK_SCDB_4		0
-+#define PXA2XX_SSP_CLK_SCDB_1		1
-+
-+extern struct snd_soc_cpu_dai pxa_ssp_dai[3];
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/Kconfig
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/Kconfig
-@@ -0,0 +1,32 @@
-+menu "SoC Audio for the Samsung S3C24XX"
-+
-+config SND_S3C24XX_SOC
-+	tristate "SoC Audio for the Samsung S3C24XX chips"
-+	depends on ARCH_S3C2410 && SND
-+	select SND_PCM
-+	help
-+	  Say Y or M if you want to add support for codecs attached to
-+	  the S3C24XX AC97, I2S or SSP interface. You will also need
-+	  to select the audio interfaces to support below.
-+
-+config SND_S3C24XX_SOC_I2S
-+	tristate
-+
-+config SND_S3C24XX_SOC_SMDK2440
-+	tristate "SoC I2S Audio support for SMDK2440"
-+	depends on SND_S3C24XX_SOC && MACH_SMDK
-+	select SND_S3C24XX_SOC_I2S
-+	select SND_SOC_UDA1380
-+	help
-+	  Say Y if you want to add support for SoC audio on SMDK2440
-+
-+config SND_S3C24XX_SOC_NEO1973_WM8753
-+	tristate "SoC I2S Audio support for NEO1973 - WM8753"
-+	depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
-+	select SND_S3C24XX_SOC_I2S
-+	select SND_SOC_WM8753
-+	help
-+	  Say Y if you want to add support for SoC audio on FIC Neo1973
-+	  with the WM8753.
-+endmenu
-+
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/Makefile
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/Makefile
-@@ -0,0 +1,13 @@
-+# S3c24XX Platform Support
-+snd-soc-s3c24xx-objs := s3c24xx-pcm.o
-+snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
-+
-+obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
-+obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
-+
-+# S3C24XX Machine Support
-+snd-soc-smdk2440-objs := smdk2440.o
-+snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
-+
-+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440) += snd-soc-smdk2440.o
-+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-i2s.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-i2s.c
-@@ -0,0 +1,439 @@
-+/*
-+ * s3c24xx-i2s.c  --  ALSA Soc Audio Layer
-+ *
-+ * (c) 2006 Wolfson Microelectronics PLC.
-+ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * (c) 2004-2005 Simtec Electronics
-+ *	http://armlinux.simtec.co.uk/
-+ *	Ben Dooks <ben at simtec.co.uk>
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *
-+ *  Revision history
-+ *    11th Dec 2006   Merged with Simtec driver
-+ *    10th Nov 2006   Initial version.
-+ */
-+
-+#include <linux/init.h>
-+#include <linux/module.h>
-+#include <linux/device.h>
-+#include <linux/delay.h>
-+#include <linux/clk.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/initval.h>
-+#include <sound/soc.h>
-+
-+#include <asm/hardware.h>
-+#include <asm/io.h>
-+#include <asm/arch/regs-iis.h>
-+#include <asm/arch/regs-gpio.h>
-+#include <asm/arch/regs-clock.h>
-+#include <asm/arch/audio.h>
-+#include <asm/dma.h>
-+#include <asm/arch/dma.h>
-+
-+#include "s3c24xx-pcm.h"
-+#include "s3c24xx-i2s.h"
-+
-+#define S3C24XX_I2S_DEBUG 0
-+#if S3C24XX_I2S_DEBUG
-+#define DBG(x...) printk(KERN_DEBUG x)
-+#else
-+#define DBG(x...)
-+#endif
-+
-+static struct s3c2410_dma_client s3c24xx_dma_client_out = {
-+	.name = "I2S PCM Stereo out"
-+};
-+
-+static struct s3c2410_dma_client s3c24xx_dma_client_in = {
-+	.name = "I2S PCM Stereo in"
-+};
-+
-+static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = {
-+	.client		= &s3c24xx_dma_client_out,
-+	.channel	= DMACH_I2S_OUT,
-+	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO
-+};
-+
-+static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = {
-+	.client		= &s3c24xx_dma_client_in,
-+	.channel	= DMACH_I2S_IN,
-+	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO
-+};
-+
-+struct s3c24xx_i2s_info {
-+	void __iomem	*regs;
-+	struct clk	*iis_clk;
-+};
-+static struct s3c24xx_i2s_info s3c24xx_i2s;
-+
-+static void s3c24xx_snd_txctrl(int on)
-+{
-+	u32 iisfcon;
-+	u32 iiscon;
-+	u32 iismod;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
-+	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
-+	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-+
-+	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+
-+	if (on) {
-+		iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
-+		iiscon  |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN;
-+		iiscon  &= ~S3C2410_IISCON_TXIDLE;
-+		iismod  |= S3C2410_IISMOD_TXMODE;
-+
-+		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
-+		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
-+		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
-+	} else {
-+		/* note, we have to disable the FIFOs otherwise bad things
-+		 * seem to happen when the DMA stops. According to the
-+		 * Samsung supplied kernel, this should allow the DMA
-+		 * engine and FIFOs to reset. If this isn't allowed, the
-+		 * DMA engine will simply freeze randomly.
-+		 */
-+
-+		iisfcon &= ~S3C2410_IISFCON_TXENABLE;
-+		iisfcon &= ~S3C2410_IISFCON_TXDMA;
-+		iiscon  |=  S3C2410_IISCON_TXIDLE;
-+		iiscon  &= ~S3C2410_IISCON_TXDMAEN;
-+		iismod  &= ~S3C2410_IISMOD_TXMODE;
-+
-+		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
-+		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
-+		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	}
-+
-+	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+}
-+
-+static void s3c24xx_snd_rxctrl(int on)
-+{
-+	u32 iisfcon;
-+	u32 iiscon;
-+	u32 iismod;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
-+	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
-+	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-+
-+	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+
-+	if (on) {
-+		iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
-+		iiscon  |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN;
-+		iiscon  &= ~S3C2410_IISCON_RXIDLE;
-+		iismod  |= S3C2410_IISMOD_RXMODE;
-+
-+		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
-+		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
-+		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
-+	} else {
-+		/* note, we have to disable the FIFOs otherwise bad things
-+		 * seem to happen when the DMA stops. According to the
-+		 * Samsung supplied kernel, this should allow the DMA
-+		 * engine and FIFOs to reset. If this isn't allowed, the
-+		 * DMA engine will simply freeze randomly.
-+		 */
-+
-+        iisfcon &= ~S3C2410_IISFCON_RXENABLE;
-+        iisfcon &= ~S3C2410_IISFCON_RXDMA;
-+        iiscon  |= S3C2410_IISCON_RXIDLE;
-+        iiscon  &= ~S3C2410_IISCON_RXDMAEN;
-+		iismod  &= ~S3C2410_IISMOD_RXMODE;
-+
-+		writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
-+		writel(iiscon,  s3c24xx_i2s.regs + S3C2410_IISCON);
-+		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	}
-+
-+	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+}
-+
-+/*
-+ * Wait for the LR signal to allow synchronisation to the L/R clock
-+ * from the codec. May only be needed for slave mode.
-+ */
-+static int s3c24xx_snd_lrsync(void)
-+{
-+	u32 iiscon;
-+	unsigned long timeout = jiffies + msecs_to_jiffies(5);
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	while (1) {
-+		iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
-+		if (iiscon & S3C2410_IISCON_LRINDEX)
-+			break;
-+
-+		if (timeout < jiffies)
-+			return -ETIMEDOUT;
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * Check whether CPU is the master or slave
-+ */
-+static inline int s3c24xx_snd_is_clkmaster(void)
-+{
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
-+}
-+
-+/*
-+ * Set S3C24xx I2S DAI format
-+ */
-+static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
-+		unsigned int fmt)
-+{
-+	u32 iismod;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	DBG("hw_params r: IISMOD: %lx \n", iismod);
-+
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		iismod |= S3C2410_IISMOD_SLAVE;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iismod |= S3C2410_IISMOD_MSB;
-+		break;
-+	case SND_SOC_DAIFMT_I2S:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	DBG("hw_params w: IISMOD: %lx \n", iismod);
-+	return 0;
-+}
-+
-+static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	u32 iismod;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-+		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
-+	else
-+		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
-+
-+	/* Working copies of register */
-+	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	DBG("hw_params r: IISMOD: %lx\n", iismod);
-+
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S8:
-+		break;
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		iismod |= S3C2410_IISMOD_16BIT;
-+		break;
-+	}
-+
-+	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	DBG("hw_params w: IISMOD: %lx\n", iismod);
-+	return 0;
-+}
-+
-+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	int ret = 0;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		if (!s3c24xx_snd_is_clkmaster()) {
-+			ret = s3c24xx_snd_lrsync();
-+			if (ret)
-+				goto exit_err;
-+		}
-+
-+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-+			s3c24xx_snd_rxctrl(1);
-+		else
-+			s3c24xx_snd_txctrl(1);
-+		break;
-+	case SNDRV_PCM_TRIGGER_STOP:
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-+			s3c24xx_snd_rxctrl(0);
-+		else
-+			s3c24xx_snd_txctrl(0);
-+		break;
-+	default:
-+		ret = -EINVAL;
-+		break;
-+	}
-+
-+exit_err:
-+	return ret;
-+}
-+
-+/*
-+ * Set S3C24xx Clock source
-+ */
-+static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
-+	int clk_id, unsigned int freq, int dir)
-+{
-+	u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	iismod &= ~S3C2440_IISMOD_MPLL;
-+
-+	switch (clk_id) {
-+	case S3C24XX_CLKSRC_PCLK:
-+		break;
-+	case S3C24XX_CLKSRC_MPLL:
-+		iismod |= S3C2440_IISMOD_MPLL;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-+	return 0;
-+}
-+
-+/*
-+ * Set S3C24xx Clock dividers
-+ */
-+static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
-+	int div_id, int div)
-+{
-+	u32 reg;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	switch (div_id) {
-+	case S3C24XX_DIV_MCLK:
-+		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK;
-+		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
-+		break;
-+	case S3C24XX_DIV_BCLK:
-+		reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS);
-+		writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
-+		break;
-+	case S3C24XX_DIV_PRESCALER:
-+		writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR);
-+		reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
-+		writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+/*
-+ * To avoid duplicating clock code, allow machine driver to
-+ * get the clockrate from here.
-+ */
-+u32 s3c24xx_i2s_get_clockrate(void)
-+{
-+	return clk_get_rate(s3c24xx_i2s.iis_clk);
-+}
-+EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
-+
-+static int s3c24xx_i2s_probe(struct platform_device *pdev)
-+{
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
-+	if (s3c24xx_i2s.regs == NULL)
-+		return -ENXIO;
-+
-+	s3c24xx_i2s.iis_clk=clk_get(&pdev->dev, "iis");
-+	if (s3c24xx_i2s.iis_clk == NULL) {
-+		DBG("failed to get iis_clock\n");
-+		return -ENODEV;
-+	}
-+	clk_enable(s3c24xx_i2s.iis_clk);
-+
-+	/* Configure the I2S pins in correct mode */
-+	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
-+	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
-+	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
-+	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
-+	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
-+
-+	writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
-+
-+	s3c24xx_snd_txctrl(0);
-+	s3c24xx_snd_rxctrl(0);
-+
-+	return 0;
-+}
-+
-+#define S3C24XX_I2S_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+struct snd_soc_cpu_dai s3c24xx_i2s_dai = {
-+	.name = "s3c24xx-i2s",
-+	.id = 0,
-+	.type = SND_SOC_DAI_I2S,
-+	.probe = s3c24xx_i2s_probe,
-+	.playback = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = S3C24XX_I2S_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-+	.capture = {
-+		.channels_min = 2,
-+		.channels_max = 2,
-+		.rates = S3C24XX_I2S_RATES,
-+		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-+	.ops = {
-+		.trigger = s3c24xx_i2s_trigger,
-+		.hw_params = s3c24xx_i2s_hw_params,},
-+	.dai_ops = {
-+		.set_fmt = s3c24xx_i2s_set_fmt,
-+		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
-+		.set_sysclk = s3c24xx_i2s_set_sysclk,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
-+
-+/* Module information */
-+MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
-+MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-i2s.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-i2s.h
-@@ -0,0 +1,35 @@
-+/*
-+ * s3c24xx-i2s.c  --  ALSA Soc Audio Layer
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Graeme Gregory
-+ *         graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    10th Nov 2006   Initial version.
-+ */
-+
-+#ifndef S3C24XXI2S_H_
-+#define S3C24XXI2S_H_
-+
-+/* clock sources */
-+#define S3C24XX_CLKSRC_PCLK 0
-+#define S3C24XX_CLKSRC_MPLL 1
-+
-+/* Clock dividers */
-+#define S3C24XX_DIV_MCLK	0
-+#define S3C24XX_DIV_BCLK	1
-+#define S3C24XX_DIV_PRESCALER	2
-+
-+/* prescaler */
-+#define S3C24XX_PRESCALE(a,b) \
-+	(((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT))
-+
-+u32 s3c24xx_i2s_get_clockrate(void);
-+
-+#endif /*S3C24XXI2S_H_*/
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-pcm.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-pcm.c
-@@ -0,0 +1,464 @@
-+/*
-+ * s3c24xx-pcm.c  --  ALSA Soc Audio Layer
-+ *
-+ * (c) 2006 Wolfson Microelectronics PLC.
-+ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * (c) 2004-2005 Simtec Electronics
-+ *	http://armlinux.simtec.co.uk/
-+ *	Ben Dooks <ben at simtec.co.uk>
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  Revision history
-+ *    11th Dec 2006   Merged with Simtec driver
-+ *    10th Nov 2006   Initial version.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/init.h>
-+#include <linux/platform_device.h>
-+#include <linux/slab.h>
-+#include <linux/dma-mapping.h>
-+
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+
-+#include <asm/dma.h>
-+#include <asm/io.h>
-+#include <asm/hardware.h>
-+#include <asm/arch/dma.h>
-+#include <asm/arch/audio.h>
-+
-+#include "s3c24xx-pcm.h"
-+
-+#define S3C24XX_PCM_DEBUG 0
-+#if S3C24XX_PCM_DEBUG
-+#define DBG(x...) printk(KERN_DEBUG x)
-+#else
-+#define DBG(x...)
-+#endif
-+
-+static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
-+	.info			= SNDRV_PCM_INFO_INTERLEAVED |
-+				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
-+				    SNDRV_PCM_INFO_MMAP |
-+				    SNDRV_PCM_INFO_MMAP_VALID,
-+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-+				    SNDRV_PCM_FMTBIT_U16_LE |
-+				    SNDRV_PCM_FMTBIT_U8 |
-+				    SNDRV_PCM_FMTBIT_S8,
-+	.channels_min		= 2,
-+	.channels_max		= 2,
-+	.buffer_bytes_max	= 128*1024,
-+	.period_bytes_min	= PAGE_SIZE,
-+	.period_bytes_max	= PAGE_SIZE*2,
-+	.periods_min		= 2,
-+	.periods_max		= 128,
-+	.fifo_size		= 32,
-+};
-+
-+struct s3c24xx_runtime_data {
-+	spinlock_t lock;
-+	int state;
-+	unsigned int dma_loaded;
-+	unsigned int dma_limit;
-+	unsigned int dma_period;
-+	dma_addr_t dma_start;
-+	dma_addr_t dma_pos;
-+	dma_addr_t dma_end;
-+	struct s3c24xx_pcm_dma_params *params;
-+};
-+
-+/* s3c24xx_pcm_enqueue
-+ *
-+ * place a dma buffer onto the queue for the dma system
-+ * to handle.
-+*/
-+static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
-+{
-+	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+	dma_addr_t pos = prtd->dma_pos;
-+	int ret;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	while (prtd->dma_loaded < prtd->dma_limit) {
-+		unsigned long len = prtd->dma_period;
-+
-+		DBG("dma_loaded: %d\n",prtd->dma_loaded);
-+
-+		if ((pos + len) > prtd->dma_end) {
-+			len  = prtd->dma_end - pos;
-+			DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
-+			       __FUNCTION__, len);
-+		}
-+
-+		ret = s3c2410_dma_enqueue(prtd->params->channel, substream, pos, len);
-+
-+		if (ret == 0) {
-+			prtd->dma_loaded++;
-+			pos += prtd->dma_period;
-+			if (pos >= prtd->dma_end)
-+				pos = prtd->dma_start;
-+		} else
-+			break;
-+	}
-+
-+	prtd->dma_pos = pos;
-+}
-+
-+static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
-+							void *dev_id, int size,
-+							enum s3c2410_dma_buffresult result)
-+{
-+	struct snd_pcm_substream *substream = dev_id;
-+	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
-+		return;
-+
-+	if (substream)
-+		snd_pcm_period_elapsed(substream);
-+
-+	spin_lock(&prtd->lock);
-+	if (prtd->state & ST_RUNNING) {
-+		prtd->dma_loaded--;
-+		s3c24xx_pcm_enqueue(substream);
-+	}
-+
-+	spin_unlock(&prtd->lock);
-+}
-+
-+static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct s3c24xx_runtime_data *prtd = runtime->private_data;
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
-+	unsigned long totbytes = params_buffer_bytes(params);
-+	int ret=0;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	/* return if this is a bufferless transfer e.g.
-+	 * codec <--> BT codec or GSM modem -- lg FIXME */
-+	if (!dma)
-+		return 0;
-+
-+	/* prepare DMA */
-+	prtd->params = dma;
-+
-+	DBG("params %p, client %p, channel %d\n", prtd->params,
-+		prtd->params->client, prtd->params->channel);
-+
-+	ret = s3c2410_dma_request(prtd->params->channel,
-+				  prtd->params->client, NULL);
-+
-+	if (ret) {
-+		DBG(KERN_ERR "failed to get dma channel\n");
-+		return ret;
-+	}
-+
-+	/* channel needs configuring for mem=>device, increment memory addr,
-+	 * sync to pclk, half-word transfers to the IIS-FIFO. */
-+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-+		s3c2410_dma_devconfig(prtd->params->channel,
-+						S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
-+						S3C2410_DISRCC_APB, prtd->params->dma_addr);
-+
-+		s3c2410_dma_config(prtd->params->channel,
-+						2, S3C2410_DCON_SYNC_PCLK | S3C2410_DCON_HANDSHAKE);
-+	} else {
-+		s3c2410_dma_config(prtd->params->channel,
-+						2, S3C2410_DCON_HANDSHAKE | S3C2410_DCON_SYNC_PCLK);
-+
-+		s3c2410_dma_devconfig(prtd->params->channel,
-+						S3C2410_DMASRC_HW, 0x3,
-+						prtd->params->dma_addr);
-+	}
-+
-+	s3c2410_dma_set_buffdone_fn(prtd->params->channel,
-+				    s3c24xx_audio_buffdone);
-+
-+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-+
-+	runtime->dma_bytes = totbytes;
-+
-+	spin_lock_irq(&prtd->lock);
-+	prtd->dma_loaded = 0;
-+	prtd->dma_limit = runtime->hw.periods_min;
-+	prtd->dma_period = params_period_bytes(params);
-+	prtd->dma_start = runtime->dma_addr;
-+	prtd->dma_pos = prtd->dma_start;
-+	prtd->dma_end = prtd->dma_start + totbytes;
-+	spin_unlock_irq(&prtd->lock);
-+
-+	return 0;
-+}
-+
-+static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
-+{
-+	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	/* TODO - do we need to ensure DMA flushed */
-+	snd_pcm_set_runtime_buffer(substream, NULL);
-+
-+	if (prtd->params) {
-+		s3c2410_dma_free(prtd->params->channel, prtd->params->client);
-+		prtd->params = NULL;
-+	}
-+
-+	return 0;
-+}
-+
-+static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
-+{
-+	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+	int ret = 0;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	/* return if this is a bufferless transfer e.g.
-+	 * codec <--> BT codec or GSM modem -- lg FIXME */
-+	if (!prtd->params)
-+	 	return 0;
-+
-+	/* flush the DMA channel */
-+	s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
-+	prtd->dma_loaded = 0;
-+	prtd->dma_pos = prtd->dma_start;
-+
-+	/* enqueue dma buffers */
-+	s3c24xx_pcm_enqueue(substream);
-+
-+	return ret;
-+}
-+
-+static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-+{
-+	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+	int ret = 0;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	spin_lock(&prtd->lock);
-+
-+	switch (cmd) {
-+	case SNDRV_PCM_TRIGGER_START:
-+	case SNDRV_PCM_TRIGGER_RESUME:
-+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-+		prtd->state |= ST_RUNNING;
-+		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
-+		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
-+		break;
-+
-+	case SNDRV_PCM_TRIGGER_STOP:
-+	case SNDRV_PCM_TRIGGER_SUSPEND:
-+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-+		prtd->state &= ~ST_RUNNING;
-+		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP);
-+		break;
-+
-+	default:
-+		ret = -EINVAL;
-+		break;
-+	}
-+
-+	spin_unlock(&prtd->lock);
-+
-+	return ret;
-+}
-+
-+static snd_pcm_uframes_t s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct s3c24xx_runtime_data *prtd = runtime->private_data;
-+	unsigned long res;
-+	dma_addr_t src, dst;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	spin_lock(&prtd->lock);
-+	s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
-+
-+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-+		res = dst - prtd->dma_start;
-+	else
-+		res = src - prtd->dma_start;
-+
-+	spin_unlock(&prtd->lock);
-+
-+	DBG("Pointer %x %x\n",src,dst);
-+
-+	/* we seem to be getting the odd error from the pcm library due
-+	 * to out-of-bounds pointers. this is maybe due to the dma engine
-+	 * not having loaded the new values for the channel before being
-+	 * callled... (todo - fix )
-+	 */
-+
-+	if (res >= snd_pcm_lib_buffer_bytes(substream)) {
-+		if (res == snd_pcm_lib_buffer_bytes(substream))
-+			res = 0;
-+	}
-+
-+	return bytes_to_frames(substream->runtime, res);
-+}
-+
-+static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct s3c24xx_runtime_data *prtd;
-+
-+	int ret;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
-+
-+	prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
-+	if (prtd == NULL)
-+		return -ENOMEM;
-+
-+	spin_lock_init(&prtd->lock);
-+
-+	runtime->private_data = prtd;
-+	return 0;
-+}
-+
-+static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+	struct s3c24xx_runtime_data *prtd = runtime->private_data;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	if (prtd)
-+		kfree(prtd);
-+	else
-+		DBG("s3c24xx_pcm_close called with prtd == NULL\n");
-+
-+	return 0;
-+}
-+
-+static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
-+	struct vm_area_struct *vma)
-+{
-+	struct snd_pcm_runtime *runtime = substream->runtime;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-+                                     runtime->dma_area,
-+                                     runtime->dma_addr,
-+                                     runtime->dma_bytes);
-+}
-+
-+static struct snd_pcm_ops s3c24xx_pcm_ops = {
-+	.open		= s3c24xx_pcm_open,
-+	.close		= s3c24xx_pcm_close,
-+	.ioctl		= snd_pcm_lib_ioctl,
-+	.hw_params	= s3c24xx_pcm_hw_params,
-+	.hw_free	= s3c24xx_pcm_hw_free,
-+	.prepare	= s3c24xx_pcm_prepare,
-+	.trigger	= s3c24xx_pcm_trigger,
-+	.pointer	= s3c24xx_pcm_pointer,
-+	.mmap		= s3c24xx_pcm_mmap,
-+};
-+
-+static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-+{
-+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-+	struct snd_dma_buffer *buf = &substream->dma_buffer;
-+	size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-+	buf->dev.dev = pcm->card->dev;
-+	buf->private_data = NULL;
-+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
-+					   &buf->addr, GFP_KERNEL);
-+	if (!buf->area)
-+		return -ENOMEM;
-+	buf->bytes = size;
-+	return 0;
-+}
-+
-+static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
-+{
-+	struct snd_pcm_substream *substream;
-+	struct snd_dma_buffer *buf;
-+	int stream;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	for (stream = 0; stream < 2; stream++) {
-+		substream = pcm->streams[stream].substream;
-+		if (!substream)
-+			continue;
-+
-+		buf = &substream->dma_buffer;
-+		if (!buf->area)
-+			continue;
-+
-+		dma_free_writecombine(pcm->card->dev, buf->bytes,
-+				      buf->area, buf->addr);
-+		buf->area = NULL;
-+	}
-+}
-+
-+static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
-+
-+static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
-+	struct snd_pcm *pcm)
-+{
-+	int ret = 0;
-+
-+	DBG("Entered %s\n", __FUNCTION__);
-+
-+	if (!card->dev->dma_mask)
-+		card->dev->dma_mask = &s3c24xx_pcm_dmamask;
-+	if (!card->dev->coherent_dma_mask)
-+		card->dev->coherent_dma_mask = 0xffffffff;
-+
-+	if (dai->playback.channels_min) {
-+		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
-+			SNDRV_PCM_STREAM_PLAYBACK);
-+		if (ret)
-+			goto out;
-+	}
-+
-+	if (dai->capture.channels_min) {
-+		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
-+			SNDRV_PCM_STREAM_CAPTURE);
-+		if (ret)
-+			goto out;
-+	}
-+ out:
-+	return ret;
-+}
-+
-+struct snd_soc_platform s3c24xx_soc_platform = {
-+	.name		= "s3c24xx-audio",
-+	.pcm_ops 	= &s3c24xx_pcm_ops,
-+	.pcm_new	= s3c24xx_pcm_new,
-+	.pcm_free	= s3c24xx_pcm_free_dma_buffers,
-+};
-+
-+EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
-+
-+MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
-+MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-pcm.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/s3c24xx-pcm.h
-@@ -0,0 +1,32 @@
-+/*
-+ *  s3c24xx-pcm.h --
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ *  ALSA PCM interface for the Samsung S3C24xx CPU
-+ */
-+
-+#ifndef _S3C24XX_PCM_H
-+#define _S3C24XX_PCM_H
-+
-+#define ST_RUNNING		(1<<0)
-+#define ST_OPENED		(1<<1)
-+
-+struct s3c24xx_pcm_dma_params {
-+	struct s3c2410_dma_client *client;			/* stream identifier */
-+	int channel;						/* Channel ID */
-+	dma_addr_t dma_addr;
-+};
-+
-+#define S3C24XX_DAI_I2S			0
-+
-+extern struct snd_soc_cpu_dai s3c24xx_i2s_dai;
-+
-+/* platform data */
-+extern struct snd_soc_platform s3c24xx_soc_platform;
-+extern struct snd_ac97_bus_ops s3c24xx_ac97_ops;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/s3c24xx/smdk2440.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/s3c24xx/smdk2440.c
-@@ -0,0 +1,318 @@
-+/*
-+ * smdk2440.c  --  ALSA Soc Audio Layer
-+ *
-+ * (c) 2006 Wolfson Microelectronics PLC.
-+ * Graeme Gregory graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * (c) 2004-2005 Simtec Electronics
-+ *	http://armlinux.simtec.co.uk/
-+ *	Ben Dooks <ben at simtec.co.uk>
-+ *
-+ *  This program is free software; you can redistribute  it and/or modify it
-+ *  under  the terms of  the GNU General  Public License as published by the
-+ *  Free Software Foundation;  either version 2 of the  License, or (at your
-+ *  option) any later version.
-+ *
-+ * This module is a modified version of the s3c24xx I2S driver supplied by
-+ * Ben Dooks of Simtec and rejigged to the ASoC style at Wolfson Microelectronics
-+ *
-+ *  Revision history
-+ *    11th Dec 2006   Merged with Simtec driver
-+ *    10th Nov 2006   Initial version.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/timer.h>
-+#include <linux/interrupt.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/mach-types.h>
-+#include <asm/hardware.h>
-+#include <asm/hardware/scoop.h>
-+#include <asm/arch/regs-iis.h>
-+#include <asm/arch/regs-clock.h>
-+#include <asm/arch/regs-gpio.h>
-+#include <asm/arch/audio.h>
-+#include <asm/io.h>
-+#include <asm/arch/spi-gpio.h>
-+#include "../codecs/uda1380.h"
-+#include "s3c24xx-pcm.h"
-+#include "s3c24xx-i2s.h"
-+
-+#define SMDK2440_DEBUG 0
-+#if SMDK2440_DEBUG
-+#define DBG(x...) printk(KERN_DEBUG x)
-+#else
-+#define DBG(x...)
-+#endif
-+
-+/* audio clock in Hz */
-+#define SMDK_CLOCK_SOURCE S3C24XX_CLKSRC_MPLL
-+#define SMDK_CRYSTAL_CLOCK 16934400
-+
-+static int smdk2440_startup(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec *codec = rtd->socdev->codec;
-+
-+	DBG("Entered smdk2440_startup\n");
-+
-+	return 0;
-+}
-+
-+static void smdk2440_shutdown(struct snd_pcm_substream *substream)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_codec *codec = rtd->socdev->codec;
-+
-+	DBG("Entered smdk2440_shutdown\n");
-+}
-+
-+static int smdk2440_hw_params(struct snd_pcm_substream *substream,
-+				struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+	unsigned long iis_clkrate;
-+	int div, div256, div384, diff256, diff384, bclk, mclk;
-+	int ret;
-+	unsigned int rate=params_rate(params);
-+
-+	DBG("Entered %s\n",__FUNCTION__);
-+
-+	iis_clkrate = s3c24xx_i2s_get_clockrate();
-+
-+	/* Using PCLK doesnt seem to suit audio particularly well on these cpu's
-+	 */
-+
-+	div256 = iis_clkrate / (rate * 256);
-+	div384 = iis_clkrate / (rate * 384);
-+
-+	if (((iis_clkrate / div256) - (rate * 256)) <
-+		((rate * 256) - (iis_clkrate / (div256 + 1)))) {
-+		diff256 = (iis_clkrate / div256) - (rate * 256);
-+	} else {
-+		div256++;
-+		diff256 = (iis_clkrate / div256) - (rate * 256);
-+	}
-+
-+	if (((iis_clkrate / div384) - (rate * 384)) <
-+		((rate * 384) - (iis_clkrate / (div384 + 1)))) {
-+		diff384 = (iis_clkrate / div384) - (rate * 384);
-+	} else {
-+		div384++;
-+		diff384 = (iis_clkrate / div384) - (rate * 384);
-+	}
-+
-+	DBG("diff256 %d, diff384 %d\n", diff256, diff384);
-+
-+	if (diff256<=diff384) {
-+		DBG("Selected 256FS\n");
-+		div = div256 - 1;
-+		bclk = S3C2410_IISMOD_256FS;
-+	} else {
-+		DBG("Selected 384FS\n");
-+		div = div384 - 1;
-+		bclk = S3C2410_IISMOD_384FS;
-+	}
-+
-+	/* set codec DAI configuration */
-+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set cpu DAI configuration */
-+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set the audio system clock for DAC and ADC */
-+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK,
-+		rate, SND_SOC_CLOCK_OUT);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set MCLK division for sample rate */
-+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS );
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set BCLK division for sample rate */
-+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, bclk);
-+	if (ret < 0)
-+		return ret;
-+
-+	/* set prescaler division for sample rate */
-+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
-+		S3C24XX_PRESCALE(div,div));
-+	if (ret < 0)
-+		return ret;
-+
-+	return 0;
-+}
-+
-+static struct snd_soc_ops smdk2440_ops = {
-+	.startup = smdk2440_startup,
-+	.shutdown = smdk2440_shutdown,
-+	.hw_params = smdk2440_hw_params,
-+};
-+
-+/* smdk2440 machine dapm widgets */
-+static const struct snd_soc_dapm_widget smdk2440_dapm_widgets[] = {
-+SND_SOC_DAPM_HP("Headphone Jack", NULL),
-+SND_SOC_DAPM_MIC("Mic Jack", NULL),
-+SND_SOC_DAPM_LINE("Line Jack", NULL),
-+};
-+
-+/* smdk2440 machine audio map (connections to the codec pins) */
-+static const char* audio_map[][3] = {
-+	/* headphone connected to  HPOUT */
-+	{"Headphone Jack", NULL, "HPOUT"},
-+
-+	/* mic is connected to MICIN (via right channel of headphone jack) */
-+	{"MICIN", NULL, "Mic Jack"},
-+	{"MICIN", NULL, "Line Jack"},
-+
-+	{NULL, NULL, NULL},
-+};
-+
-+/*
-+ * Logic for a UDA1341 as attached to SMDK2440
-+ */
-+static int smdk2440_uda1341_init(struct snd_soc_codec *codec)
-+{
-+	int i, err;
-+
-+	DBG("Staring smdk2440 init\n");
-+
-+	/* Add smdk2440 specific widgets */
-+	for(i = 0; i < ARRAY_SIZE(smdk2440_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &smdk2440_dapm_widgets[i]);
-+	}
-+
-+	/* Set up smdk2440 specific audio path audio_mapnects */
-+	for(i = 0; audio_map[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-+			audio_map[i][1], audio_map[i][2]);
-+	}
-+
-+	snd_soc_dapm_sync_endpoints(codec);
-+
-+	DBG("Ending smdk2440 init\n");
-+
-+	return 0;
-+}
-+
-+/* s3c24xx digital audio interface glue - connects codec <--> CPU */
-+static struct snd_soc_dai_link s3c24xx_dai = {
-+	.name = "WM8731",
-+	.stream_name = "WM8731",
-+	.cpu_dai = &s3c24xx_i2s_dai,
-+	.codec_dai = uda1380_dai,
-+	.init = smdk2440_uda1341_init,
-+	.ops = &smdk2440_ops,
-+};
-+
-+/* smdk2440 audio machine driver */
-+static struct snd_soc_machine snd_soc_machine_smdk2440 = {
-+	.name = "SMDK2440",
-+	.dai_link = &s3c24xx_dai,
-+	.num_links = 1,
-+};
-+
-+static struct uda1380_setup_data smdk2440_uda1380_setup = {
-+	.i2c_address = 0x00,
-+};
-+
-+/* s3c24xx audio subsystem */
-+static struct snd_soc_device s3c24xx_snd_devdata = {
-+	.machine = &snd_soc_machine_smdk2440,
-+	.platform = &s3c24xx_soc_platform,
-+	.codec_dev = &soc_codec_dev_uda1380,
-+	.codec_data = &smdk2440_uda1380_setup,
-+};
-+
-+static struct platform_device *s3c24xx_snd_device;
-+
-+struct smdk2440_spi_device {
-+	struct device *dev;
-+};
-+
-+static struct smdk2440_spi_device smdk2440_spi_devdata = {
-+};
-+
-+struct s3c2410_spigpio_info smdk2440_spi_devinfo = {
-+	.pin_clk = S3C2410_GPF4,
-+	.pin_mosi = S3C2410_GPF5,
-+	.pin_miso = S3C2410_GPF6,
-+	//.board_size,
-+	//.board_info,
-+	.chip_select=NULL,
-+};
-+
-+static struct platform_device *smdk2440_spi_device;
-+
-+static int __init smdk2440_init(void)
-+{
-+	int ret;
-+
-+	if (!machine_is_smdk2440() && !machine_is_s3c2440()) {
-+		DBG("%d\n",machine_arch_type);
-+		DBG("Not a SMDK2440\n");
-+		return -ENODEV;
-+	}
-+
-+	s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
-+	if (!s3c24xx_snd_device) {
-+		DBG("platform_dev_alloc failed\n");
-+		return -ENOMEM;
-+	}
-+
-+	platform_set_drvdata(s3c24xx_snd_device, &s3c24xx_snd_devdata);
-+	s3c24xx_snd_devdata.dev = &s3c24xx_snd_device->dev;
-+	ret = platform_device_add(s3c24xx_snd_device);
-+
-+	if (ret)
-+		platform_device_put(s3c24xx_snd_device);
-+
-+	// Create a bitbanged SPI device
-+
-+	smdk2440_spi_device = platform_device_alloc("s3c24xx-spi-gpio",-1);
-+	if (!smdk2440_spi_device) {
-+		DBG("smdk2440_spi_device : platform_dev_alloc failed\n");
-+		return -ENOMEM;
-+	}
-+	DBG("Return Code %d\n",ret);
-+
-+	platform_set_drvdata(smdk2440_spi_device, &smdk2440_spi_devdata);
-+	smdk2440_spi_devdata.dev = &smdk2440_spi_device->dev;
-+	smdk2440_spi_devdata.dev->platform_data = &smdk2440_spi_devinfo;
-+	ret = platform_device_add(smdk2440_spi_device);
-+
-+	if (ret)
-+		platform_device_put(smdk2440_spi_device);
-+
-+	return ret;
-+}
-+
-+static void __exit smdk2440_exit(void)
-+{
-+	platform_device_unregister(s3c24xx_snd_device);
-+}
-+
-+module_init(smdk2440_init);
-+module_exit(smdk2440_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
-+MODULE_DESCRIPTION("ALSA SoC SMDK2440");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8956.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8956.c
-@@ -0,0 +1,724 @@
-+/*
-+ * wm8956.c  --  WM8956 ALSA SoC Audio driver
-+ *
-+ * Author: Liam Girdwood
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8956.h"
-+
-+#define AUDIO_NAME "wm8956"
-+#define WM8956_VERSION "0.2"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8956_DEBUG 0
-+
-+#ifdef WM8956_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8956;
-+
-+/*
-+ * wm8956 register cache
-+ * We can't read the WM8956 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8956_reg[WM8956_CACHEREGNUM] = {
-+	0x0097, 0x0097, 0x0000, 0x0000,
-+	0x0000, 0x0008, 0x0000, 0x000a,
-+	0x01c0, 0x0000, 0x00ff, 0x00ff,
-+	0x0000, 0x0000, 0x0000, 0x0000, //r15
-+	0x0000, 0x007b, 0x0100, 0x0032,
-+	0x0000, 0x00c3, 0x00c3, 0x01c0,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000, //r31
-+	0x0100, 0x0100, 0x0050, 0x0050,
-+	0x0050, 0x0050, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0040, 0x0000,
-+	0x0000, 0x0050, 0x0050, 0x0000, //47
-+	0x0002, 0x0037, 0x004d, 0x0080,
-+	0x0008, 0x0031, 0x0026, 0x00e9,
-+};
-+
-+/*
-+ * read wm8956 register cache
-+ */
-+static inline unsigned int wm8956_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8956_RESET)
-+		return 0;
-+	if (reg >= WM8956_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8956 register cache
-+ */
-+static inline void wm8956_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8956_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8956 register space
-+ */
-+static int wm8956_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8956 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8956_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8956_reset(c)	wm8956_write(c, WM8956_RESET, 0)
-+
-+/* todo - complete enumerated controls */
-+static const char *wm8956_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-+
-+static const struct soc_enum wm8956_enum[] = {
-+	SOC_ENUM_SINGLE(WM8956_DACCTL1, 1, 4, wm8956_deemph),
-+};
-+
-+/* to complete */
-+static const struct snd_kcontrol_new wm8956_snd_controls[] = {
-+
-+SOC_DOUBLE_R("Headphone Playback Volume", WM8956_LOUT1, WM8956_ROUT1,
-+	0, 127, 0),
-+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8956_LOUT1, WM8956_ROUT1,
-+	7, 1, 0),
-+SOC_DOUBLE_R("PCM Volume", WM8956_LDAC, WM8956_RDAC,
-+	0, 127, 0),
-+
-+SOC_ENUM("Playback De-emphasis", wm8956_enum[0]),
-+};
-+
-+/* add non dapm controls */
-+static int wm8956_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8956_snd_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8956_snd_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Left Output Mixer */
-+static const struct snd_kcontrol_new wm8956_loutput_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8956_LOUTMIX1, 8, 1, 0),
-+};
-+
-+/* Right Output Mixer */
-+static const struct snd_kcontrol_new wm8956_routput_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8956_ROUTMIX2, 8, 1, 0),
-+};
-+
-+static const struct snd_soc_dapm_widget wm8956_dapm_widgets[] = {
-+SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8956_loutput_mixer_controls[0],
-+	ARRAY_SIZE(wm8956_loutput_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8956_loutput_mixer_controls[0],
-+	ARRAY_SIZE(wm8956_routput_mixer_controls)),
-+};
-+
-+static const char *intercon[][3] = {
-+	/* TODO */
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8956_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8956_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8956_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, intercon[i][0],
-+			intercon[i][1], intercon[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+static int wm8956_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
-+		unsigned int fmt)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 iface = 0;
-+
-+	/* set master/slave audio interface */
-+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-+	case SND_SOC_DAIFMT_CBM_CFM:
-+		iface |= 0x0040;
-+		break;
-+	case SND_SOC_DAIFMT_CBS_CFS:
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* interface format */
-+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-+	case SND_SOC_DAIFMT_I2S:
-+		iface |= 0x0002;
-+		break;
-+	case SND_SOC_DAIFMT_RIGHT_J:
-+		break;
-+	case SND_SOC_DAIFMT_LEFT_J:
-+		iface |= 0x0001;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_A:
-+		iface |= 0x0003;
-+		break;
-+	case SND_SOC_DAIFMT_DSP_B:
-+		iface |= 0x0013;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* clock inversion */
-+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-+	case SND_SOC_DAIFMT_NB_NF:
-+		break;
-+	case SND_SOC_DAIFMT_IB_IF:
-+		iface |= 0x0090;
-+		break;
-+	case SND_SOC_DAIFMT_IB_NF:
-+		iface |= 0x0080;
-+		break;
-+	case SND_SOC_DAIFMT_NB_IF:
-+		iface |= 0x0010;
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	/* set iface */
-+	wm8956_write(codec, WM8956_IFACE1, iface);
-+	return 0;
-+}
-+
-+static int wm8956_hw_params(struct snd_pcm_substream *substream,
-+	struct snd_pcm_hw_params *params)
-+{
-+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+	struct snd_soc_device *socdev = rtd->socdev;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	u16 iface = wm8956_read_reg_cache(codec, WM8956_IFACE1) & 0xfff3;
-+
-+	/* bit size */
-+	switch (params_format(params)) {
-+	case SNDRV_PCM_FORMAT_S16_LE:
-+		break;
-+	case SNDRV_PCM_FORMAT_S20_3LE:
-+		iface |= 0x0004;
-+		break;
-+	case SNDRV_PCM_FORMAT_S24_LE:
-+		iface |= 0x0008;
-+		break;
-+	}
-+
-+	/* set iface */
-+	wm8956_write(codec, WM8956_IFACE1, iface);
-+	return 0;
-+}
-+
-+static int wm8956_mute(struct snd_soc_codec_dai *dai, int mute)
-+{
-+	struct snd_soc_codec *codec = dai->codec;
-+	u16 mute_reg = wm8956_read_reg_cache(codec, WM8956_DACCTL1) & 0xfff7;
-+
-+	if (mute)
-+		wm8956_write(codec, WM8956_DACCTL1, mute_reg | 0x8);
-+	else
-+		wm8956_write(codec, WM8956_DACCTL1, mute_reg);
-+	return 0;
-+}
-+
-+static int wm8956_dapm_event(struct snd_soc_codec *codec, int event)
-+{
-+#if 0
-+	switch (event) {
-+	case SNDRV_CTL_POWER_D0: /* full On */
-+		/* vref/mid, osc on, dac unmute */
-+
-+		break;
-+	case SNDRV_CTL_POWER_D1: /* partial On */
-+	case SNDRV_CTL_POWER_D2: /* partial On */
-+		break;
-+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+		/* everything off except vref/vmid, */
-+		break;
-+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+		/* everything off, dac mute, inactive */
-+		break;
-+	}
-+#endif
-+	// tmp
-+	wm8956_write(codec, WM8956_POWER1, 0xffff);
-+	wm8956_write(codec, WM8956_POWER2, 0xffff);
-+	wm8956_write(codec, WM8956_POWER3, 0xffff);
-+	codec->dapm_state = event;
-+	return 0;
-+}
-+
-+/* PLL divisors */
-+struct _pll_div {
-+	u32 pre_div:1;
-+	u32 n:4;
-+	u32 k:24;
-+};
-+
-+static struct _pll_div pll_div;
-+
-+/* The size in bits of the pll divide multiplied by 10
-+ * to allow rounding later */
-+#define FIXED_PLL_SIZE ((1 << 24) * 10)
-+
-+static void pll_factors(unsigned int target, unsigned int source)
-+{
-+	unsigned long long Kpart;
-+	unsigned int K, Ndiv, Nmod;
-+
-+	Ndiv = target / source;
-+	if (Ndiv < 6) {
-+		source >>= 1;
-+		pll_div.pre_div = 1;
-+		Ndiv = target / source;
-+	} else
-+		pll_div.pre_div = 0;
-+
-+	if ((Ndiv < 6) || (Ndiv > 12))
-+		printk(KERN_WARNING
-+			"WM8956 N value outwith recommended range! N = %d\n",Ndiv);
-+
-+	pll_div.n = Ndiv;
-+	Nmod = target % source;
-+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
-+
-+	do_div(Kpart, source);
-+
-+	K = Kpart & 0xFFFFFFFF;
-+
-+	/* Check if we need to round */
-+	if ((K % 10) >= 5)
-+		K += 5;
-+
-+	/* Move down to proper range now rounding is done */
-+	K /= 10;
-+
-+	pll_div.k = K;
-+}
-+
-+static int wm8956_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
-+		int pll_id, unsigned int freq_in, unsigned int freq_out)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+	int found = 0;
-+#if 0
-+	if (freq_in == 0 || freq_out == 0) {
-+		/* disable the pll */
-+		/* turn PLL power off */
-+	}
-+#endif
-+
-+	pll_factors(freq_out * 8, freq_in);
-+
-+	if (!found)
-+		return -EINVAL;
-+
-+	reg = wm8956_read_reg_cache(codec, WM8956_PLLN) & 0x1e0;
-+	wm8956_write(codec, WM8956_PLLN, reg | (pll_div.pre_div << 4)
-+		| pll_div.n);
-+	wm8956_write(codec, WM8956_PLLK1, pll_div.k >> 16 );
-+	wm8956_write(codec, WM8956_PLLK2, (pll_div.k >> 8) & 0xff);
-+	wm8956_write(codec, WM8956_PLLK3, pll_div.k &0xff);
-+	wm8956_write(codec, WM8956_CLOCK1, 4);
-+
-+	return 0;
-+}
-+
-+static int wm8956_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
-+		int div_id, int div)
-+{
-+	struct snd_soc_codec *codec = codec_dai->codec;
-+	u16 reg;
-+
-+	switch (div_id) {
-+	case WM8956_SYSCLKSEL:
-+		reg = wm8956_read_reg_cache(codec, WM8956_CLOCK1) & 0x1fe;
-+		wm8956_write(codec, WM8956_CLOCK1, reg | div);
-+		break;
-+	case WM8956_SYSCLKDIV:
-+		reg = wm8956_read_reg_cache(codec, WM8956_CLOCK1) & 0x1f9;
-+		wm8956_write(codec, WM8956_CLOCK1, reg | div);
-+		break;
-+	case WM8956_DACDIV:
-+		reg = wm8956_read_reg_cache(codec, WM8956_CLOCK1) & 0x1c7;
-+		wm8956_write(codec, WM8956_CLOCK1, reg | div);
-+		break;
-+	case WM8956_OPCLKDIV:
-+		reg = wm8956_read_reg_cache(codec, WM8956_PLLN) & 0x03f;
-+		wm8956_write(codec, WM8956_PLLN, reg | div);
-+		break;
-+	case WM8956_DCLKDIV:
-+		reg = wm8956_read_reg_cache(codec, WM8956_CLOCK2) & 0x03f;
-+		wm8956_write(codec, WM8956_CLOCK2, reg | div);
-+		break;
-+	case WM8956_TOCLKSEL:
-+		reg = wm8956_read_reg_cache(codec, WM8956_ADDCTL1) & 0x1fd;
-+		wm8956_write(codec, WM8956_ADDCTL1, reg | div);
-+		break;
-+	default:
-+		return -EINVAL;
-+	}
-+
-+	return 0;
-+}
-+
-+#define WM8956_RATES \
-+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-+
-+#define WM8956_FORMATS \
-+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
-+	SNDRV_PCM_FMTBIT_S24_LE)
-+
-+struct snd_soc_codec_dai wm8956_dai = {
-+	.name = "WM8956",
-+	.playback = {
-+		.stream_name = "Playback",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8956_RATES,
-+		.formats = WM8956_FORMATS,},
-+	.capture = {
-+		.stream_name = "Capture",
-+		.channels_min = 1,
-+		.channels_max = 2,
-+		.rates = WM8956_RATES,
-+		.formats = WM8956_FORMATS,},
-+	.ops = {
-+		.hw_params = wm8956_hw_params,
-+	},
-+	.dai_ops = {
-+		.digital_mute = wm8956_mute,
-+		.set_fmt = wm8956_set_dai_fmt,
-+		.set_clkdiv = wm8956_set_dai_clkdiv,
-+		.set_pll = wm8956_set_dai_pll,
-+	},
-+};
-+EXPORT_SYMBOL_GPL(wm8956_dai);
-+
-+
-+/* To complete PM */
-+static int wm8956_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	wm8956_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+	return 0;
-+}
-+
-+static int wm8956_resume(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int i;
-+	u8 data[2];
-+	u16 *cache = codec->reg_cache;
-+
-+	/* Sync reg_cache with the hardware */
-+	for (i = 0; i < ARRAY_SIZE(wm8956_reg); i++) {
-+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-+		data[1] = cache[i] & 0x00ff;
-+		codec->hw_write(codec->control_data, data, 2);
-+	}
-+	wm8956_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+	wm8956_dapm_event(codec, codec->suspend_dapm_state);
-+	return 0;
-+}
-+
-+/*
-+ * initialise the WM8956 driver
-+ * register the mixer and dsp interfaces with the kernel
-+ */
-+static int wm8956_init(struct snd_soc_device *socdev)
-+{
-+	struct snd_soc_codec *codec = socdev->codec;
-+	int reg, ret = 0;
-+
-+	codec->name = "WM8956";
-+	codec->owner = THIS_MODULE;
-+	codec->read = wm8956_read_reg_cache;
-+	codec->write = wm8956_write;
-+	codec->dapm_event = wm8956_dapm_event;
-+	codec->dai = &wm8956_dai;
-+	codec->num_dai = 1;
-+	codec->reg_cache_size = ARRAY_SIZE(wm8956_reg);
-+
-+	codec->reg_cache =
-+			kzalloc(sizeof(u16) * ARRAY_SIZE(wm8956_reg), GFP_KERNEL);
-+	if (codec->reg_cache == NULL)
-+		return -ENOMEM;
-+	memcpy(codec->reg_cache,
-+		wm8956_reg, sizeof(u16) * ARRAY_SIZE(wm8956_reg));
-+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8956_reg);
-+
-+	wm8956_reset(codec);
-+
-+	/* register pcms */
-+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-+	if (ret < 0) {
-+		printk(KERN_ERR "wm8956: failed to create pcms\n");
-+		goto pcm_err;
-+	}
-+
-+	/* power on device */
-+	wm8956_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-+
-+	/*  set the update bits */
-+	reg = wm8956_read_reg_cache(codec, WM8956_LOUT1);
-+	wm8956_write(codec, WM8956_LOUT1, reg | 0x0100);
-+	reg = wm8956_read_reg_cache(codec, WM8956_ROUT1);
-+	wm8956_write(codec, WM8956_ROUT1, reg | 0x0100);
-+
-+	wm8956_add_controls(codec);
-+	wm8956_add_widgets(codec);
-+	ret = snd_soc_register_card(socdev);
-+	if (ret < 0) {
-+      	printk(KERN_ERR "wm8956: failed to register card\n");
-+		goto card_err;
-+    }
-+	return ret;
-+
-+card_err:
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+pcm_err:
-+	kfree(codec->reg_cache);
-+	return ret;
-+}
-+
-+static struct snd_soc_device *wm8956_socdev;
-+
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+
-+/*
-+ * WM8956 2 wire address is 0x1a
-+ */
-+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-+
-+/* Magic definition of all other variables and things */
-+I2C_CLIENT_INSMOD;
-+
-+static struct i2c_driver wm8956_i2c_driver;
-+static struct i2c_client client_template;
-+
-+/* If the i2c layer weren't so broken, we could pass this kind of data
-+   around */
-+
-+static int wm8956_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-+{
-+	struct snd_soc_device *socdev = wm8956_socdev;
-+	struct wm8956_setup_data *setup = socdev->codec_data;
-+	struct snd_soc_codec *codec = socdev->codec;
-+	struct i2c_client *i2c;
-+	int ret;
-+
-+	if (addr != setup->i2c_address)
-+		return -ENODEV;
-+
-+	client_template.adapter = adap;
-+	client_template.addr = addr;
-+
-+	i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
-+	if (i2c == NULL) {
-+		kfree(codec);
-+		return -ENOMEM;
-+	}
-+	memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+	i2c_set_clientdata(i2c, codec);
-+	codec->control_data = i2c;
-+
-+	ret = i2c_attach_client(i2c);
-+	if (ret < 0) {
-+		err("failed to attach codec at addr %x\n", addr);
-+		goto err;
-+	}
-+
-+	ret = wm8956_init(socdev);
-+	if (ret < 0) {
-+		err("failed to initialise WM8956\n");
-+		goto err;
-+	}
-+	return ret;
-+
-+err:
-+	kfree(codec);
-+	kfree(i2c);
-+	return ret;
-+}
-+
-+static int wm8956_i2c_detach(struct i2c_client *client)
-+{
-+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
-+	i2c_detach_client(client);
-+	kfree(codec->reg_cache);
-+	kfree(client);
-+	return 0;
-+}
-+
-+static int wm8956_i2c_attach(struct i2c_adapter *adap)
-+{
-+	return i2c_probe(adap, &addr_data, wm8956_codec_probe);
-+}
-+
-+// tmp
-+#define I2C_DRIVERID_WM8956 0xfefe
-+
-+/* corgi i2c codec control layer */
-+static struct i2c_driver wm8956_i2c_driver = {
-+	.driver = {
-+		.name = "WM8956 I2C Codec",
-+		.owner = THIS_MODULE,
-+	},
-+	.id =             I2C_DRIVERID_WM8956,
-+	.attach_adapter = wm8956_i2c_attach,
-+	.detach_client =  wm8956_i2c_detach,
-+	.command =        NULL,
-+};
-+
-+static struct i2c_client client_template = {
-+	.name =   "WM8956",
-+	.driver = &wm8956_i2c_driver,
-+};
-+#endif
-+
-+static int wm8956_probe(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct wm8956_setup_data *setup;
-+	struct snd_soc_codec *codec;
-+	int ret = 0;
-+
-+	info("WM8956 Audio Codec %s", WM8956_VERSION);
-+
-+	setup = socdev->codec_data;
-+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-+	if (codec == NULL)
-+		return -ENOMEM;
-+
-+	socdev->codec = codec;
-+	mutex_init(&codec->mutex);
-+	INIT_LIST_HEAD(&codec->dapm_widgets);
-+	INIT_LIST_HEAD(&codec->dapm_paths);
-+
-+	wm8956_socdev = socdev;
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	if (setup->i2c_address) {
-+		normal_i2c[0] = setup->i2c_address;
-+		codec->hw_write = (hw_write_t)i2c_master_send;
-+		ret = i2c_add_driver(&wm8956_i2c_driver);
-+		if (ret != 0)
-+			printk(KERN_ERR "can't add i2c driver");
-+	}
-+#else
-+	/* Add other interfaces here */
-+#endif
-+	return ret;
-+}
-+
-+/* power down chip */
-+static int wm8956_remove(struct platform_device *pdev)
-+{
-+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-+	struct snd_soc_codec *codec = socdev->codec;
-+
-+	if (codec->control_data)
-+		wm8956_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+
-+	snd_soc_free_pcms(socdev);
-+	snd_soc_dapm_free(socdev);
-+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
-+	i2c_del_driver(&wm8956_i2c_driver);
-+#endif
-+	kfree(codec);
-+
-+	return 0;
-+}
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8956 = {
-+	.probe = 	wm8956_probe,
-+	.remove = 	wm8956_remove,
-+	.suspend = 	wm8956_suspend,
-+	.resume =	wm8956_resume,
-+};
-+
-+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8956);
-+
-+MODULE_DESCRIPTION("ASoC WM8956 driver");
-+MODULE_AUTHOR("Liam Girdwood");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8956.h
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8956.h
-@@ -0,0 +1,116 @@
-+/*
-+ * wm8956.h  --  WM8956 Soc Audio driver
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#ifndef _WM8956_H
-+#define _WM8956_H
-+
-+/* WM8956 register space */
-+
-+
-+#define WM8956_CACHEREGNUM 	56
-+
-+#define WM8956_LINVOL		0x0
-+#define WM8956_RINVOL	0x1
-+#define WM8956_LOUT1		0x2
-+#define WM8956_ROUT1		0x3
-+#define WM8956_CLOCK1	0x4
-+#define WM8956_DACCTL1	0x5
-+#define WM8956_DACCTL2	0x6
-+#define WM8956_IFACE1		0x7
-+#define WM8956_CLOCK2	0x8
-+#define WM8956_IFACE2		0x9
-+#define WM8956_LDAC		0xa
-+#define WM8956_RDAC		0xb
-+
-+#define WM8956_RESET		0xf
-+#define WM8956_3D				0x10
-+
-+#define WM8956_ADDCTL1		0x17
-+#define WM8956_ADDCTL2		0x18
-+#define WM8956_POWER1	0x19
-+#define WM8956_POWER2	0x1a
-+#define WM8956_ADDCTL3	0x1b
-+#define WM8956_APOP1		0x1c
-+#define WM8956_APOP2		0x1d
-+
-+#define WM8956_LINPATH	0x20
-+#define WM8956_RINPATH	0x21
-+#define WM8956_LOUTMIX1	0x22
-+
-+#define WM8956_ROUTMIX2	0x25
-+#define WM8956_MONOMIX1	0x26
-+#define WM8956_MONOMIX2	0x27
-+#define WM8956_LOUT2		0x28
-+#define WM8956_ROUT2		0x29
-+#define WM8956_MONO		0x2a
-+#define WM8956_INBMIX1	0x2b
-+#define WM8956_INBMIX2	0x2c
-+#define WM8956_BYPASS1	0x2d
-+#define WM8956_BYPASS2	0x2e
-+#define WM8956_POWER3	0x2f
-+#define WM8956_ADDCTL4		0x30
-+#define WM8956_CLASSD1		0x31
-+
-+#define WM8956_CLASSD3		0x33
-+#define WM8956_PLLN		0x34
-+#define WM8956_PLLK1		0x35
-+#define WM8956_PLLK2		0x36
-+#define WM8956_PLLK3		0x37
-+
-+
-+/*
-+ * WM8956 Clock dividers
-+ */
-+#define WM8956_SYSCLKDIV 		0
-+#define WM8956_DACDIV			1
-+#define WM8956_OPCLKDIV			2
-+#define WM8956_DCLKDIV			3
-+#define WM8956_TOCLKSEL			4
-+#define WM8956_SYSCLKSEL		5
-+
-+#define WM8956_SYSCLK_DIV_1		(0 << 1)
-+#define WM8956_SYSCLK_DIV_2		(2 << 1)
-+
-+#define WM8956_SYSCLK_MCLK		(0 << 0)
-+#define WM8956_SYSCLK_PLL		(1 << 0)
-+
-+#define WM8956_DAC_DIV_1		(0 << 3)
-+#define WM8956_DAC_DIV_1_5		(1 << 3)
-+#define WM8956_DAC_DIV_2		(2 << 3)
-+#define WM8956_DAC_DIV_3		(3 << 3)
-+#define WM8956_DAC_DIV_4		(4 << 3)
-+#define WM8956_DAC_DIV_5_5		(5 << 3)
-+#define WM8956_DAC_DIV_6		(6 << 3)
-+
-+#define WM8956_DCLK_DIV_1_5		(0 << 6)
-+#define WM8956_DCLK_DIV_2		(1 << 6)
-+#define WM8956_DCLK_DIV_3		(2 << 6)
-+#define WM8956_DCLK_DIV_4		(3 << 6)
-+#define WM8956_DCLK_DIV_6		(4 << 6)
-+#define WM8956_DCLK_DIV_8		(5 << 6)
-+#define WM8956_DCLK_DIV_12		(6 << 6)
-+#define WM8956_DCLK_DIV_16		(7 << 6)
-+
-+#define WM8956_TOCLK_F19		(0 << 1)
-+#define WM8956_TOCLK_F21		(1 << 1)
-+
-+#define WM8956_OPCLK_DIV_1		(0 << 0)
-+#define WM8956_OPCLK_DIV_2		(1 << 0)
-+#define WM8956_OPCLK_DIV_3		(2 << 0)
-+#define WM8956_OPCLK_DIV_4		(3 << 0)
-+#define WM8956_OPCLK_DIV_5_5	(4 << 0)
-+#define WM8956_OPCLK_DIV_6		(5 << 0)
-+
-+struct wm8956_setup_data {
-+	unsigned short i2c_address;
-+};
-+
-+extern struct snd_soc_codec_dai wm8956_dai;
-+extern struct snd_soc_codec_device soc_codec_dev_wm8956;
-+
-+#endif
-Index: linux-2.6.21-moko/sound/soc/codecs/wm8960.c
-===================================================================
---- /dev/null
-+++ linux-2.6.21-moko/sound/soc/codecs/wm8960.c
-@@ -0,0 +1,766 @@
-+/*
-+ * wm8960.c  --  WM8960 ALSA SoC Audio driver
-+ *
-+ * Author: Liam Girdwood
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License version 2 as
-+ * published by the Free Software Foundation.
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/init.h>
-+#include <linux/delay.h>
-+#include <linux/pm.h>
-+#include <linux/i2c.h>
-+#include <linux/platform_device.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/pcm_params.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+#include <sound/initval.h>
-+
-+#include "wm8960.h"
-+
-+#define AUDIO_NAME "wm8960"
-+#define WM8960_VERSION "0.1"
-+
-+/*
-+ * Debug
-+ */
-+
-+#define WM8960_DEBUG 0
-+
-+#ifdef WM8960_DEBUG
-+#define dbg(format, arg...) \
-+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-+#else
-+#define dbg(format, arg...) do {} while (0)
-+#endif
-+#define err(format, arg...) \
-+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-+#define info(format, arg...) \
-+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-+#define warn(format, arg...) \
-+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-+
-+struct snd_soc_codec_device soc_codec_dev_wm8960;
-+
-+/*
-+ * wm8960 register cache
-+ * We can't read the WM8960 register space when we are
-+ * using 2 wire for device control, so we cache them instead.
-+ */
-+static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
-+	0x0097, 0x0097, 0x0000, 0x0000,
-+	0x0000, 0x0008, 0x0000, 0x000a,
-+	0x01c0, 0x0000, 0x00ff, 0x00ff,
-+	0x0000, 0x0000, 0x0000, 0x0000, //r15
-+	0x0000, 0x007b, 0x0100, 0x0032,
-+	0x0000, 0x00c3, 0x00c3, 0x01c0,
-+	0x0000, 0x0000, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0000, 0x0000, //r31
-+	0x0100, 0x0100, 0x0050, 0x0050,
-+	0x0050, 0x0050, 0x0000, 0x0000,
-+	0x0000, 0x0000, 0x0040, 0x0000,
-+	0x0000, 0x0050, 0x0050, 0x0000, //47
-+	0x0002, 0x0037, 0x004d, 0x0080,
-+	0x0008, 0x0031, 0x0026, 0x00e9,
-+};
-+
-+/*
-+ * read wm8960 register cache
-+ */
-+static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
-+	unsigned int reg)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg == WM8960_RESET)
-+		return 0;
-+	if (reg >= WM8960_CACHEREGNUM)
-+		return -1;
-+	return cache[reg];
-+}
-+
-+/*
-+ * write wm8960 register cache
-+ */
-+static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
-+	u16 reg, unsigned int value)
-+{
-+	u16 *cache = codec->reg_cache;
-+	if (reg >= WM8960_CACHEREGNUM)
-+		return;
-+	cache[reg] = value;
-+}
-+
-+/*
-+ * write to the WM8960 register space
-+ */
-+static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
-+	unsigned int value)
-+{
-+	u8 data[2];
-+
-+	/* data is
-+	 *   D15..D9 WM8960 register offset
-+	 *   D8...D0 register data
-+	 */
-+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
-+	data[1] = value & 0x00ff;
-+
-+	wm8960_write_reg_cache (codec, reg, value);
-+	if (codec->hw_write(codec->control_data, data, 2) == 2)
-+		return 0;
-+	else
-+		return -EIO;
-+}
-+
-+#define wm8960_reset(c)	wm8960_write(c, WM8960_RESET, 0)
-+
-+/* enumerated controls */
-+static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-+static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
-+	"Right Inverted", "Stereo Inversion"};
-+static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
-+static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
-+static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
-+static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
-+
-+static const struct soc_enum wm8960_enum[] = {
-+	SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
-+	SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
-+	SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
-+	SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
-+	SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
-+	SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
-+	SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
-+};
-+
-+/* to complete */
-+static const struct snd_kcontrol_new wm8960_snd_controls[] = {
-+SOC_DOUBLE_R("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
-+	0, 63, 0),
-+SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
-+	6, 1, 0),
-+SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
-+	7, 1, 0),
-+SOC_DOUBLE_R("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
-+	0, 127, 0),
-+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
-+	7, 1, 0),
-+SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
-+SOC_ENUM("ADC Polarity", wm8960_enum[1]),
-+SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
-+SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
-+
-+SOC_ENUM("DAC Polarity", wm8960_enum[2]),
-+
-+SOC_DOUBLE_R("PCM Volume", WM8960_LDAC, WM8960_RDAC,
-+	0, 127, 0),
-+
-+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
-+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
-+SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
-+SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
-+
-+SOC_ENUM("ALC Function", wm8960_enum[5]),
-+SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
-+SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
-+SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
-+SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
-+SOC_ENUM("ALC Mode", wm8960_enum[6]),
-+SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
-+SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
-+
-+SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
-+SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
-+
-+SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
-+	0, 127, 0),
-+};
-+
-+/* add non dapm controls */
-+static int wm8960_add_controls(struct snd_soc_codec *codec)
-+{
-+	int err, i;
-+
-+	for (i = 0; i < ARRAY_SIZE(wm8960_snd_controls); i++) {
-+		if ((err = snd_ctl_add(codec->card,
-+				snd_soc_cnew(&wm8960_snd_controls[i],codec, NULL))) < 0)
-+			return err;
-+	}
-+
-+	return 0;
-+}
-+
-+/* Left Output Mixer */
-+static const struct snd_kcontrol_new wm8960_loutput_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8960_LOUTMIX1, 8, 1, 0),
-+};
-+
-+/* Right Output Mixer */
-+static const struct snd_kcontrol_new wm8960_routput_mixer_controls[] = {
-+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8960_ROUTMIX2, 8, 1, 0),
-+};
-+
-+static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
-+SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8960_loutput_mixer_controls[0],
-+	ARRAY_SIZE(wm8960_loutput_mixer_controls)),
-+SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
-+	&wm8960_loutput_mixer_controls[0],
-+	ARRAY_SIZE(wm8960_routput_mixer_controls)),
-+};
-+
-+static const char *intercon[][3] = {
-+	/* TODO */
-+	/* terminator */
-+	{NULL, NULL, NULL},
-+};
-+
-+static int wm8960_add_widgets(struct snd_soc_codec *codec)
-+{
-+	int i;
-+
-+	for(i = 0; i < ARRAY_SIZE(wm8960_dapm_widgets); i++) {
-+		snd_soc_dapm_new_control(codec, &wm8960_dapm_widgets[i]);
-+	}
-+
-+	/* set up audio path interconnects */
-+	for(i = 0; intercon[i][0] != NULL; i++) {
-+		snd_soc_dapm_connect_input(codec, intercon[i][0],
-+			intercon[i][1], intercon[i][2]);
-+	}
-+
-+	snd_soc_dapm_new_widgets(codec);
-+	return 0;
-+}
-+
-+static int wm8960_set_dai_fmt(stru