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(¶ms, _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, ® );
+ 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, ® );
+ 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, ® );
+- 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, ® );
+ 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, ® );
+ 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&quo