r3047 - branches/src/target/kernel/2.6.23.x/patches
shoragan at sita.openmoko.org
shoragan at sita.openmoko.org
Wed Sep 26 18:16:04 CEST 2007
Author: shoragan
Date: 2007-09-26 18:16:04 +0200 (Wed, 26 Sep 2007)
New Revision: 3047
Removed:
branches/src/target/kernel/2.6.23.x/patches/alsa-2.6.23-rc1-commit.diff
branches/src/target/kernel/2.6.23.x/patches/s3c2410_udc_from_upstream.patch
Modified:
branches/src/target/kernel/2.6.23.x/patches/series
Log:
Drop patches which have been merged
Deleted: branches/src/target/kernel/2.6.23.x/patches/alsa-2.6.23-rc1-commit.diff
===================================================================
--- branches/src/target/kernel/2.6.23.x/patches/alsa-2.6.23-rc1-commit.diff 2007-09-26 16:12:59 UTC (rev 3046)
+++ branches/src/target/kernel/2.6.23.x/patches/alsa-2.6.23-rc1-commit.diff 2007-09-26 16:16:04 UTC (rev 3047)
@@ -1,10960 +0,0 @@
---- 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/2.6.23.x/patches/s3c2410_udc_from_upstream.patch
===================================================================
--- branches/src/target/kernel/2.6.23.x/patches/s3c2410_udc_from_upstream.patch 2007-09-26 16:12:59 UTC (rev 3046)
+++ branches/src/target/kernel/2.6.23.x/patches/s3c2410_udc_from_upstream.patch 2007-09-26 16:16:04 UTC (rev 3047)
@@ -1,2265 +0,0 @@
-From: Arnaud Patard <arnaud.patard at rtp-net.org>
-Date: Thu, 7 Jun 2007 04:05:49 +0000 (-0700)
-Subject: USB Gadget driver for Samsung s3c2410 ARM SoC
-X-Git-Tag: v2.6.23-rc1~1083^2~86
-X-Git-Url: http://git.kernel.org/?p=linux%2Fkernel%2Fgit%2Ftorvalds%2Flinux-2.6.git;a=commitdiff_plain;h=3fc154b6b8134b98bb94d60cad9a46ec1ffbe372;hp=7a4eb7fd50d4df99fc1f623e6d90680d9fca3d82
-
-USB Gadget driver for Samsung s3c2410 ARM SoC
-
-This patch adds the support for the Usb Device Controller on Samsung
-S3C24xx SoCs. This driver passes all tests from testusb (including #13)
-and has been tested on S3C2410, S3C24212, and S3C2440 SoCs.
-
-Whitespace updates, minor cleanups by David
-
-Signed-off-by: Arnaud Patard <arnaud.patard at rtp-net.org>
-Signed-off-by: Ben Dooks <ben-linux at fluff.org>
-Cc: Herbert Pötzl <herbert at 13thfloor.at>
-Signed-off-by: David Brownell <dbrownell at users.sourceforge.net>
-Signed-off-by: Greg Kroah-Hartman <gregkh at suse.de>
----
-
-diff --git a/drivers/usb/gadget/Kconfig b/drivers/usb/gadget/Kconfig
-index 0576888..74eaa7d 100644
---- a/drivers/usb/gadget/Kconfig
-+++ b/drivers/usb/gadget/Kconfig
-@@ -208,6 +208,27 @@ config USB_OTG
-
- Select this only if your OMAP board has a Mini-AB connector.
-
-+config USB_GADGET_S3C2410
-+ boolean "S3C2410 USB Device Controller"
-+ depends on ARCH_S3C2410
-+ help
-+ Samsung's S3C2410 is an ARM-4 processor with an integrated
-+ full speed USB 1.1 device controller. It has 4 configurable
-+ endpoints, as well as endpoint zero (for control transfers).
-+
-+ This driver has been tested on the S3C2410, S3C2412, and
-+ S3C2440 processors.
-+
-+config USB_S3C2410
-+ tristate
-+ depends on USB_GADGET_S3C2410
-+ default USB_GADGET
-+ select USB_GADGET_SELECTED
-+
-+config USB_S3C2410_DEBUG
-+ boolean "S3C2410 udc debug messages"
-+ depends on USB_GADGET_S3C2410
-+
- config USB_GADGET_AT91
- boolean "AT91 USB Device Port"
- depends on ARCH_AT91 && !ARCH_AT91SAM9RL
-diff --git a/drivers/usb/gadget/Makefile b/drivers/usb/gadget/Makefile
-index 2d41e84..bff2783 100644
---- a/drivers/usb/gadget/Makefile
-+++ b/drivers/usb/gadget/Makefile
-@@ -7,6 +7,7 @@ obj-$(CONFIG_USB_PXA2XX) += pxa2xx_udc.o
- obj-$(CONFIG_USB_GOKU) += goku_udc.o
- obj-$(CONFIG_USB_OMAP) += omap_udc.o
- obj-$(CONFIG_USB_LH7A40X) += lh7a40x_udc.o
-+obj-$(CONFIG_USB_S3C2410) += s3c2410_udc.o
- obj-$(CONFIG_USB_AT91) += at91_udc.o
- obj-$(CONFIG_USB_FSL_USB2) += fsl_usb2_udc.o
- obj-$(CONFIG_USB_M66592) += m66592-udc.o
-diff --git a/drivers/usb/gadget/s3c2410_udc.c b/drivers/usb/gadget/s3c2410_udc.c
-new file mode 100644
-index 0000000..d60748a
---- /dev/null
-+++ b/drivers/usb/gadget/s3c2410_udc.c
-@@ -0,0 +1,2078 @@
-+/*
-+ * linux/drivers/usb/gadget/s3c2410_udc.c
-+ *
-+ * Samsung S3C24xx series on-chip full speed USB device controllers
-+ *
-+ * Copyright (C) 2004-2007 Herbert Pötzl - Arnaud Patard
-+ * Additional cleanups by Ben Dooks <ben-linux at fluff.org>
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License as published by
-+ * the Free Software Foundation; either version 2 of the License, or
-+ * (at your option) any later version.
-+ *
-+ * This 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/module.h>
-+#include <linux/kernel.h>
-+#include <linux/delay.h>
-+#include <linux/ioport.h>
-+#include <linux/sched.h>
-+#include <linux/slab.h>
-+#include <linux/smp_lock.h>
-+#include <linux/errno.h>
-+#include <linux/init.h>
-+#include <linux/timer.h>
-+#include <linux/list.h>
-+#include <linux/interrupt.h>
-+#include <linux/platform_device.h>
-+#include <linux/version.h>
-+#include <linux/clk.h>
-+
-+#include <linux/debugfs.h>
-+#include <linux/seq_file.h>
-+
-+#include <linux/usb.h>
-+#include <linux/usb_gadget.h>
-+
-+#include <asm/byteorder.h>
-+#include <asm/io.h>
-+#include <asm/irq.h>
-+#include <asm/system.h>
-+#include <asm/unaligned.h>
-+#include <asm/arch/irqs.h>
-+
-+#include <asm/arch/hardware.h>
-+#include <asm/arch/regs-clock.h>
-+#include <asm/arch/regs-gpio.h>
-+#include <asm/arch/regs-udc.h>
-+#include <asm/arch/udc.h>
-+
-+#include <asm/mach-types.h>
-+
-+#include "s3c2410_udc.h"
-+
-+#define DRIVER_DESC "S3C2410 USB Device Controller Gadget"
-+#define DRIVER_VERSION "29 Apr 2007"
-+#define DRIVER_AUTHOR "Herbert Pötzl <herbert at 13thfloor.at>, " \
-+ "Arnaud Patard <arnaud.patard at rtp-net.org>"
-+
-+static const char gadget_name[] = "s3c2410_udc";
-+static const char driver_desc[] = DRIVER_DESC;
-+
-+static struct s3c2410_udc *the_controller;
-+static struct clk *udc_clock;
-+static struct clk *usb_bus_clock;
-+static void __iomem *base_addr;
-+static u64 rsrc_start;
-+static u64 rsrc_len;
-+static struct dentry *s3c2410_udc_debugfs_root;
-+
-+static inline u32 udc_read(u32 reg)
-+{
-+ return readb(base_addr + reg);
-+}
-+
-+static inline void udc_write(u32 value, u32 reg)
-+{
-+ writeb(value, base_addr + reg);
-+}
-+
-+static inline void udc_writeb(void __iomem *base, u32 value, u32 reg)
-+{
-+ writeb(value, base + reg);
-+}
-+
-+static struct s3c2410_udc_mach_info *udc_info;
-+
-+/*************************** DEBUG FUNCTION ***************************/
-+#define DEBUG_NORMAL 1
-+#define DEBUG_VERBOSE 2
-+
-+#ifdef CONFIG_USB_S3C2410_DEBUG
-+#define USB_S3C2410_DEBUG_LEVEL 0
-+
-+static uint32_t s3c2410_ticks = 0;
-+
-+static int dprintk(int level, const char *fmt, ...)
-+{
-+ static char printk_buf[1024];
-+ static long prevticks;
-+ static int invocation;
-+ va_list args;
-+ int len;
-+
-+ if (level > USB_S3C2410_DEBUG_LEVEL)
-+ return 0;
-+
-+ if (s3c2410_ticks != prevticks) {
-+ prevticks = s3c2410_ticks;
-+ invocation = 0;
-+ }
-+
-+ len = scnprintf(printk_buf,
-+ sizeof(printk_buf), "%1lu.%02d USB: ",
-+ prevticks, invocation++);
-+
-+ va_start(args, fmt);
-+ len = vscnprintf(printk_buf+len,
-+ sizeof(printk_buf)-len, fmt, args);
-+ va_end(args);
-+
-+ return printk(KERN_DEBUG "%s", printk_buf);
-+}
-+#else
-+static int dprintk(int level, const char *fmt, ...)
-+{
-+ return 0;
-+}
-+#endif
-+static int s3c2410_udc_debugfs_seq_show(struct seq_file *m, void *p)
-+{
-+ u32 addr_reg,pwr_reg,ep_int_reg,usb_int_reg;
-+ u32 ep_int_en_reg, usb_int_en_reg, ep0_csr;
-+ u32 ep1_i_csr1,ep1_i_csr2,ep1_o_csr1,ep1_o_csr2;
-+ u32 ep2_i_csr1,ep2_i_csr2,ep2_o_csr1,ep2_o_csr2;
-+
-+ addr_reg = udc_read(S3C2410_UDC_FUNC_ADDR_REG);
-+ pwr_reg = udc_read(S3C2410_UDC_PWR_REG);
-+ ep_int_reg = udc_read(S3C2410_UDC_EP_INT_REG);
-+ usb_int_reg = udc_read(S3C2410_UDC_USB_INT_REG);
-+ ep_int_en_reg = udc_read(S3C2410_UDC_EP_INT_EN_REG);
-+ usb_int_en_reg = udc_read(S3C2410_UDC_USB_INT_EN_REG);
-+ udc_write(0, S3C2410_UDC_INDEX_REG);
-+ ep0_csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ udc_write(1, S3C2410_UDC_INDEX_REG);
-+ ep1_i_csr1 = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ ep1_i_csr2 = udc_read(S3C2410_UDC_IN_CSR2_REG);
-+ ep1_o_csr1 = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ ep1_o_csr2 = udc_read(S3C2410_UDC_IN_CSR2_REG);
-+ udc_write(2, S3C2410_UDC_INDEX_REG);
-+ ep2_i_csr1 = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ ep2_i_csr2 = udc_read(S3C2410_UDC_IN_CSR2_REG);
-+ ep2_o_csr1 = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ ep2_o_csr2 = udc_read(S3C2410_UDC_IN_CSR2_REG);
-+
-+ seq_printf(m, "FUNC_ADDR_REG : 0x%04X\n"
-+ "PWR_REG : 0x%04X\n"
-+ "EP_INT_REG : 0x%04X\n"
-+ "USB_INT_REG : 0x%04X\n"
-+ "EP_INT_EN_REG : 0x%04X\n"
-+ "USB_INT_EN_REG : 0x%04X\n"
-+ "EP0_CSR : 0x%04X\n"
-+ "EP1_I_CSR1 : 0x%04X\n"
-+ "EP1_I_CSR2 : 0x%04X\n"
-+ "EP1_O_CSR1 : 0x%04X\n"
-+ "EP1_O_CSR2 : 0x%04X\n"
-+ "EP2_I_CSR1 : 0x%04X\n"
-+ "EP2_I_CSR2 : 0x%04X\n"
-+ "EP2_O_CSR1 : 0x%04X\n"
-+ "EP2_O_CSR2 : 0x%04X\n",
-+ addr_reg,pwr_reg,ep_int_reg,usb_int_reg,
-+ ep_int_en_reg, usb_int_en_reg, ep0_csr,
-+ ep1_i_csr1,ep1_i_csr2,ep1_o_csr1,ep1_o_csr2,
-+ ep2_i_csr1,ep2_i_csr2,ep2_o_csr1,ep2_o_csr2
-+ );
-+
-+ return 0;
-+}
-+
-+static int s3c2410_udc_debugfs_fops_open(struct inode *inode,
-+ struct file *file)
-+{
-+ return single_open(file, s3c2410_udc_debugfs_seq_show, NULL);
-+}
-+
-+static const struct file_operations s3c2410_udc_debugfs_fops = {
-+ .open = s3c2410_udc_debugfs_fops_open,
-+ .read = seq_read,
-+ .llseek = seq_lseek,
-+ .release = single_release,
-+ .owner = THIS_MODULE,
-+};
-+
-+/* io macros */
-+
-+static inline void s3c2410_udc_clear_ep0_opr(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, S3C2410_UDC_EP0_CSR_SOPKTRDY,
-+ S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_clear_ep0_sst(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ writeb(0x00, base + S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_clear_ep0_se(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, S3C2410_UDC_EP0_CSR_SSE, S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_set_ep0_ipr(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, S3C2410_UDC_EP0_CSR_IPKRDY, S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_set_ep0_de(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, S3C2410_UDC_EP0_CSR_DE, S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+inline void s3c2410_udc_set_ep0_ss(void __iomem *b)
-+{
-+ udc_writeb(b, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(b, S3C2410_UDC_EP0_CSR_SENDSTL, S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_set_ep0_de_out(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+
-+ udc_writeb(base,(S3C2410_UDC_EP0_CSR_SOPKTRDY
-+ | S3C2410_UDC_EP0_CSR_DE),
-+ S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_set_ep0_sse_out(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, (S3C2410_UDC_EP0_CSR_SOPKTRDY
-+ | S3C2410_UDC_EP0_CSR_SSE),
-+ S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+static inline void s3c2410_udc_set_ep0_de_in(void __iomem *base)
-+{
-+ udc_writeb(base, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ udc_writeb(base, (S3C2410_UDC_EP0_CSR_IPKRDY
-+ | S3C2410_UDC_EP0_CSR_DE),
-+ S3C2410_UDC_EP0_CSR_REG);
-+}
-+
-+/*------------------------- I/O ----------------------------------*/
-+
-+/*
-+ * s3c2410_udc_done
-+ */
-+static void s3c2410_udc_done(struct s3c2410_ep *ep,
-+ struct s3c2410_request *req, int status)
-+{
-+ unsigned halted = ep->halted;
-+
-+ list_del_init(&req->queue);
-+
-+ if (likely (req->req.status == -EINPROGRESS))
-+ req->req.status = status;
-+ else
-+ status = req->req.status;
-+
-+ ep->halted = 1;
-+ req->req.complete(&ep->ep, &req->req);
-+ ep->halted = halted;
-+}
-+
-+static void s3c2410_udc_nuke(struct s3c2410_udc *udc,
-+ struct s3c2410_ep *ep, int status)
-+{
-+ /* Sanity check */
-+ if (&ep->queue == NULL)
-+ return;
-+
-+ while (!list_empty (&ep->queue)) {
-+ struct s3c2410_request *req;
-+ req = list_entry (ep->queue.next, struct s3c2410_request,
-+ queue);
-+ s3c2410_udc_done(ep, req, status);
-+ }
-+}
-+
-+static inline void s3c2410_udc_clear_ep_state(struct s3c2410_udc *dev)
-+{
-+ unsigned i;
-+
-+ /* hardware SET_{CONFIGURATION,INTERFACE} automagic resets endpoint
-+ * fifos, and pending transactions mustn't be continued in any case.
-+ */
-+
-+ for (i = 1; i < S3C2410_ENDPOINTS; i++)
-+ s3c2410_udc_nuke(dev, &dev->ep[i], -ECONNABORTED);
-+}
-+
-+static inline int s3c2410_udc_fifo_count_out(void)
-+{
-+ int tmp;
-+
-+ tmp = udc_read(S3C2410_UDC_OUT_FIFO_CNT2_REG) << 8;
-+ tmp |= udc_read(S3C2410_UDC_OUT_FIFO_CNT1_REG);
-+ return tmp;
-+}
-+
-+/*
-+ * s3c2410_udc_write_packet
-+ */
-+static inline int s3c2410_udc_write_packet(int fifo,
-+ struct s3c2410_request *req,
-+ unsigned max)
-+{
-+ unsigned len = min(req->req.length - req->req.actual, max);
-+ u8 *buf = req->req.buf + req->req.actual;
-+
-+ prefetch(buf);
-+
-+ dprintk(DEBUG_VERBOSE, "%s %d %d %d %d\n", __func__,
-+ req->req.actual, req->req.length, len, req->req.actual + len);
-+
-+ req->req.actual += len;
-+
-+ udelay(5);
-+ writesb(base_addr + fifo, buf, len);
-+ return len;
-+}
-+
-+/*
-+ * s3c2410_udc_write_fifo
-+ *
-+ * return: 0 = still running, 1 = completed, negative = errno
-+ */
-+static int s3c2410_udc_write_fifo(struct s3c2410_ep *ep,
-+ struct s3c2410_request *req)
-+{
-+ unsigned count;
-+ int is_last;
-+ u32 idx;
-+ int fifo_reg;
-+ u32 ep_csr;
-+
-+ idx = ep->bEndpointAddress & 0x7F;
-+ switch (idx) {
-+ default:
-+ idx = 0;
-+ case 0:
-+ fifo_reg = S3C2410_UDC_EP0_FIFO_REG;
-+ break;
-+ case 1:
-+ fifo_reg = S3C2410_UDC_EP1_FIFO_REG;
-+ break;
-+ case 2:
-+ fifo_reg = S3C2410_UDC_EP2_FIFO_REG;
-+ break;
-+ case 3:
-+ fifo_reg = S3C2410_UDC_EP3_FIFO_REG;
-+ break;
-+ case 4:
-+ fifo_reg = S3C2410_UDC_EP4_FIFO_REG;
-+ break;
-+ }
-+
-+ count = s3c2410_udc_write_packet(fifo_reg, req, ep->ep.maxpacket);
-+
-+ /* last packet is often short (sometimes a zlp) */
-+ if (count != ep->ep.maxpacket)
-+ is_last = 1;
-+ else if (req->req.length != req->req.actual || req->req.zero)
-+ is_last = 0;
-+ else
-+ is_last = 2;
-+
-+ /* Only ep0 debug messages are interesting */
-+ if (idx == 0)
-+ dprintk(DEBUG_NORMAL,
-+ "Written ep%d %d.%d of %d b [last %d,z %d]\n",
-+ idx, count, req->req.actual, req->req.length,
-+ is_last, req->req.zero);
-+
-+ if (is_last) {
-+ /* The order is important. It prevents sending 2 packets
-+ * at the same time */
-+
-+ if (idx == 0) {
-+ /* Reset signal => no need to say 'data sent' */
-+ if (! (udc_read(S3C2410_UDC_USB_INT_REG)
-+ & S3C2410_UDC_USBINT_RESET))
-+ s3c2410_udc_set_ep0_de_in(base_addr);
-+ ep->dev->ep0state=EP0_IDLE;
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr | S3C2410_UDC_ICSR1_PKTRDY,
-+ S3C2410_UDC_IN_CSR1_REG);
-+ }
-+
-+ s3c2410_udc_done(ep, req, 0);
-+ is_last = 1;
-+ } else {
-+ if (idx == 0) {
-+ /* Reset signal => no need to say 'data sent' */
-+ if (! (udc_read(S3C2410_UDC_USB_INT_REG)
-+ & S3C2410_UDC_USBINT_RESET))
-+ s3c2410_udc_set_ep0_ipr(base_addr);
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr | S3C2410_UDC_ICSR1_PKTRDY,
-+ S3C2410_UDC_IN_CSR1_REG);
-+ }
-+ }
-+
-+ return is_last;
-+}
-+
-+static inline int s3c2410_udc_read_packet(int fifo, u8 *buf,
-+ struct s3c2410_request *req, unsigned avail)
-+{
-+ unsigned len;
-+
-+ len = min(req->req.length - req->req.actual, avail);
-+ req->req.actual += len;
-+
-+ readsb(fifo + base_addr, buf, len);
-+ return len;
-+}
-+
-+/*
-+ * return: 0 = still running, 1 = queue empty, negative = errno
-+ */
-+static int s3c2410_udc_read_fifo(struct s3c2410_ep *ep,
-+ struct s3c2410_request *req)
-+{
-+ u8 *buf;
-+ u32 ep_csr;
-+ unsigned bufferspace;
-+ int is_last=1;
-+ unsigned avail;
-+ int fifo_count = 0;
-+ u32 idx;
-+ int fifo_reg;
-+
-+ idx = ep->bEndpointAddress & 0x7F;
-+
-+ switch (idx) {
-+ default:
-+ idx = 0;
-+ case 0:
-+ fifo_reg = S3C2410_UDC_EP0_FIFO_REG;
-+ break;
-+ case 1:
-+ fifo_reg = S3C2410_UDC_EP1_FIFO_REG;
-+ break;
-+ case 2:
-+ fifo_reg = S3C2410_UDC_EP2_FIFO_REG;
-+ break;
-+ case 3:
-+ fifo_reg = S3C2410_UDC_EP3_FIFO_REG;
-+ break;
-+ case 4:
-+ fifo_reg = S3C2410_UDC_EP4_FIFO_REG;
-+ break;
-+ }
-+
-+ if (!req->req.length)
-+ return 1;
-+
-+ buf = req->req.buf + req->req.actual;
-+ bufferspace = req->req.length - req->req.actual;
-+ if (!bufferspace) {
-+ dprintk(DEBUG_NORMAL, "%s: buffer full!\n", __func__);
-+ return -1;
-+ }
-+
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+
-+ fifo_count = s3c2410_udc_fifo_count_out();
-+ dprintk(DEBUG_NORMAL, "%s fifo count : %d\n", __func__, fifo_count);
-+
-+ if (fifo_count > ep->ep.maxpacket)
-+ avail = ep->ep.maxpacket;
-+ else
-+ avail = fifo_count;
-+
-+ fifo_count = s3c2410_udc_read_packet(fifo_reg, buf, req, avail);
-+
-+ /* checking this with ep0 is not accurate as we already
-+ * read a control request
-+ **/
-+ if (idx != 0 && fifo_count < ep->ep.maxpacket) {
-+ is_last = 1;
-+ /* overflowed this request? flush extra data */
-+ if (fifo_count != avail)
-+ req->req.status = -EOVERFLOW;
-+ } else {
-+ is_last = (req->req.length <= req->req.actual) ? 1 : 0;
-+ }
-+
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ fifo_count = s3c2410_udc_fifo_count_out();
-+
-+ /* Only ep0 debug messages are interesting */
-+ if (idx == 0)
-+ dprintk(DEBUG_VERBOSE, "%s fifo count : %d [last %d]\n",
-+ __func__, fifo_count,is_last);
-+
-+ if (is_last) {
-+ if (idx == 0) {
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ ep->dev->ep0state = EP0_IDLE;
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read(S3C2410_UDC_OUT_CSR1_REG);
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr & ~S3C2410_UDC_OCSR1_PKTRDY,
-+ S3C2410_UDC_OUT_CSR1_REG);
-+ }
-+
-+ s3c2410_udc_done(ep, req, 0);
-+ } else {
-+ if (idx == 0) {
-+ s3c2410_udc_clear_ep0_opr(base_addr);
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read(S3C2410_UDC_OUT_CSR1_REG);
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr & ~S3C2410_UDC_OCSR1_PKTRDY,
-+ S3C2410_UDC_OUT_CSR1_REG);
-+ }
-+ }
-+
-+ return is_last;
-+}
-+
-+static int s3c2410_udc_read_fifo_crq(struct usb_ctrlrequest *crq)
-+{
-+ unsigned char *outbuf = (unsigned char*)crq;
-+ int bytes_read = 0;
-+
-+ udc_write(0, S3C2410_UDC_INDEX_REG);
-+
-+ bytes_read = s3c2410_udc_fifo_count_out();
-+
-+ dprintk(DEBUG_NORMAL, "%s: fifo_count=%d\n", __func__, bytes_read);
-+
-+ if (bytes_read > sizeof(struct usb_ctrlrequest))
-+ bytes_read = sizeof(struct usb_ctrlrequest);
-+
-+ readsb(S3C2410_UDC_EP0_FIFO_REG + base_addr, outbuf, bytes_read);
-+
-+ dprintk(DEBUG_VERBOSE, "%s: len=%d %02x:%02x {%x,%x,%x}\n", __func__,
-+ bytes_read, crq->bRequest, crq->bRequestType,
-+ crq->wValue, crq->wIndex, crq->wLength);
-+
-+ return bytes_read;
-+}
-+
-+static int s3c2410_udc_get_status(struct s3c2410_udc *dev,
-+ struct usb_ctrlrequest *crq)
-+{
-+ u16 status = 0;
-+ u8 ep_num = crq->wIndex & 0x7F;
-+ u8 is_in = crq->wIndex & USB_DIR_IN;
-+
-+ switch (crq->bRequestType & USB_RECIP_MASK) {
-+ case USB_RECIP_INTERFACE:
-+ break;
-+
-+ case USB_RECIP_DEVICE:
-+ status = dev->devstatus;
-+ break;
-+
-+ case USB_RECIP_ENDPOINT:
-+ if (ep_num > 4 || crq->wLength > 2)
-+ return 1;
-+
-+ if (ep_num == 0) {
-+ udc_write(0, S3C2410_UDC_INDEX_REG);
-+ status = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ status = status & S3C2410_UDC_EP0_CSR_SENDSTL;
-+ } else {
-+ udc_write(ep_num, S3C2410_UDC_INDEX_REG);
-+ if (is_in) {
-+ status = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ status = status & S3C2410_UDC_ICSR1_SENDSTL;
-+ } else {
-+ status = udc_read(S3C2410_UDC_OUT_CSR1_REG);
-+ status = status & S3C2410_UDC_OCSR1_SENDSTL;
-+ }
-+ }
-+
-+ status = status ? 1 : 0;
-+ break;
-+
-+ default:
-+ return 1;
-+ }
-+
-+ /* Seems to be needed to get it working. ouch :( */
-+ udelay(5);
-+ udc_write(status & 0xFF, S3C2410_UDC_EP0_FIFO_REG);
-+ udc_write(status >> 8, S3C2410_UDC_EP0_FIFO_REG);
-+ s3c2410_udc_set_ep0_de_in(base_addr);
-+
-+ return 0;
-+}
-+/*------------------------- usb state machine -------------------------------*/
-+static int s3c2410_udc_set_halt(struct usb_ep *_ep, int value);
-+
-+static void s3c2410_udc_handle_ep0_idle(struct s3c2410_udc *dev,
-+ struct s3c2410_ep *ep,
-+ struct usb_ctrlrequest *crq,
-+ u32 ep0csr)
-+{
-+ int len, ret, tmp;
-+
-+ /* start control request? */
-+ if (!(ep0csr & S3C2410_UDC_EP0_CSR_OPKRDY))
-+ return;
-+
-+ s3c2410_udc_nuke(dev, ep, -EPROTO);
-+
-+ len = s3c2410_udc_read_fifo_crq(crq);
-+ if (len != sizeof(*crq)) {
-+ dprintk(DEBUG_NORMAL, "setup begin: fifo READ ERROR"
-+ " wanted %d bytes got %d. Stalling out...\n",
-+ sizeof(*crq), len);
-+ s3c2410_udc_set_ep0_ss(base_addr);
-+ return;
-+ }
-+
-+ dprintk(DEBUG_NORMAL, "bRequest = %d bRequestType %d wLength = %d\n",
-+ crq->bRequest, crq->bRequestType, crq->wLength);
-+
-+ /* cope with automagic for some standard requests. */
-+ dev->req_std = (crq->bRequestType & USB_TYPE_MASK)
-+ == USB_TYPE_STANDARD;
-+ dev->req_config = 0;
-+ dev->req_pending = 1;
-+
-+ switch (crq->bRequest) {
-+ case USB_REQ_SET_CONFIGURATION:
-+ dprintk(DEBUG_NORMAL, "USB_REQ_SET_CONFIGURATION ... \n");
-+
-+ if (crq->bRequestType == USB_RECIP_DEVICE) {
-+ dev->req_config = 1;
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ }
-+ break;
-+
-+ case USB_REQ_SET_INTERFACE:
-+ dprintk(DEBUG_NORMAL, "USB_REQ_SET_INTERFACE ... \n");
-+
-+ if (crq->bRequestType == USB_RECIP_INTERFACE) {
-+ dev->req_config = 1;
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ }
-+ break;
-+
-+ case USB_REQ_SET_ADDRESS:
-+ dprintk(DEBUG_NORMAL, "USB_REQ_SET_ADDRESS ... \n");
-+
-+ if (crq->bRequestType == USB_RECIP_DEVICE) {
-+ tmp = crq->wValue & 0x7F;
-+ dev->address = tmp;
-+ udc_write((tmp | S3C2410_UDC_FUNCADDR_UPDATE),
-+ S3C2410_UDC_FUNC_ADDR_REG);
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ return;
-+ }
-+ break;
-+
-+ case USB_REQ_GET_STATUS:
-+ dprintk(DEBUG_NORMAL, "USB_REQ_GET_STATUS ... \n");
-+ s3c2410_udc_clear_ep0_opr(base_addr);
-+
-+ if (dev->req_std) {
-+ if (!s3c2410_udc_get_status(dev, crq)) {
-+ return;
-+ }
-+ }
-+ break;
-+
-+ case USB_REQ_CLEAR_FEATURE:
-+ s3c2410_udc_clear_ep0_opr(base_addr);
-+
-+ if (crq->bRequestType != USB_RECIP_ENDPOINT)
-+ break;
-+
-+ if (crq->wValue != USB_ENDPOINT_HALT || crq->wLength != 0)
-+ break;
-+
-+ s3c2410_udc_set_halt(&dev->ep[crq->wIndex & 0x7f].ep, 0);
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ return;
-+
-+ case USB_REQ_SET_FEATURE:
-+ s3c2410_udc_clear_ep0_opr(base_addr);
-+
-+ if (crq->bRequestType != USB_RECIP_ENDPOINT)
-+ break;
-+
-+ if (crq->wValue != USB_ENDPOINT_HALT || crq->wLength != 0)
-+ break;
-+
-+ s3c2410_udc_set_halt(&dev->ep[crq->wIndex & 0x7f].ep, 1);
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ return;
-+
-+ default:
-+ s3c2410_udc_clear_ep0_opr(base_addr);
-+ break;
-+ }
-+
-+ if (crq->bRequestType & USB_DIR_IN)
-+ dev->ep0state = EP0_IN_DATA_PHASE;
-+ else
-+ dev->ep0state = EP0_OUT_DATA_PHASE;
-+
-+ ret = dev->driver->setup(&dev->gadget, crq);
-+ if (ret < 0) {
-+ if (dev->req_config) {
-+ dprintk(DEBUG_NORMAL, "config change %02x fail %d?\n",
-+ crq->bRequest, ret);
-+ return;
-+ }
-+
-+ if (ret == -EOPNOTSUPP)
-+ dprintk(DEBUG_NORMAL, "Operation not supported\n");
-+ else
-+ dprintk(DEBUG_NORMAL,
-+ "dev->driver->setup failed. (%d)\n", ret);
-+
-+ udelay(5);
-+ s3c2410_udc_set_ep0_ss(base_addr);
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ dev->ep0state = EP0_IDLE;
-+ /* deferred i/o == no response yet */
-+ } else if (dev->req_pending) {
-+ dprintk(DEBUG_VERBOSE, "dev->req_pending... what now?\n");
-+ dev->req_pending=0;
-+ }
-+
-+ dprintk(DEBUG_VERBOSE, "ep0state %s\n", ep0states[dev->ep0state]);
-+}
-+
-+static void s3c2410_udc_handle_ep0(struct s3c2410_udc *dev)
-+{
-+ u32 ep0csr;
-+ struct s3c2410_ep *ep = &dev->ep[0];
-+ struct s3c2410_request *req;
-+ struct usb_ctrlrequest crq;
-+
-+ if (list_empty(&ep->queue))
-+ req = NULL;
-+ else
-+ req = list_entry(ep->queue.next, struct s3c2410_request, queue);
-+
-+ /* We make the assumption that S3C2410_UDC_IN_CSR1_REG equal to
-+ * S3C2410_UDC_EP0_CSR_REG when index is zero */
-+
-+ udc_write(0, S3C2410_UDC_INDEX_REG);
-+ ep0csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+
-+ dprintk(DEBUG_NORMAL, "ep0csr %x ep0state %s\n",
-+ ep0csr, ep0states[dev->ep0state]);
-+
-+ /* clear stall status */
-+ if (ep0csr & S3C2410_UDC_EP0_CSR_SENTSTL) {
-+ s3c2410_udc_nuke(dev, ep, -EPIPE);
-+ dprintk(DEBUG_NORMAL, "... clear SENT_STALL ...\n");
-+ s3c2410_udc_clear_ep0_sst(base_addr);
-+ dev->ep0state = EP0_IDLE;
-+ return;
-+ }
-+
-+ /* clear setup end */
-+ if (ep0csr & S3C2410_UDC_EP0_CSR_SE) {
-+ dprintk(DEBUG_NORMAL, "... serviced SETUP_END ...\n");
-+ s3c2410_udc_nuke(dev, ep, 0);
-+ s3c2410_udc_clear_ep0_se(base_addr);
-+ dev->ep0state = EP0_IDLE;
-+ }
-+
-+ switch (dev->ep0state) {
-+ case EP0_IDLE:
-+ s3c2410_udc_handle_ep0_idle(dev, ep, &crq, ep0csr);
-+ break;
-+
-+ case EP0_IN_DATA_PHASE: /* GET_DESCRIPTOR etc */
-+ dprintk(DEBUG_NORMAL, "EP0_IN_DATA_PHASE ... what now?\n");
-+ if (!(ep0csr & S3C2410_UDC_EP0_CSR_IPKRDY) && req) {
-+ s3c2410_udc_write_fifo(ep, req);
-+ }
-+ break;
-+
-+ case EP0_OUT_DATA_PHASE: /* SET_DESCRIPTOR etc */
-+ dprintk(DEBUG_NORMAL, "EP0_OUT_DATA_PHASE ... what now?\n");
-+ if ((ep0csr & S3C2410_UDC_EP0_CSR_OPKRDY) && req ) {
-+ s3c2410_udc_read_fifo(ep,req);
-+ }
-+ break;
-+
-+ case EP0_END_XFER:
-+ dprintk(DEBUG_NORMAL, "EP0_END_XFER ... what now?\n");
-+ dev->ep0state = EP0_IDLE;
-+ break;
-+
-+ case EP0_STALL:
-+ dprintk(DEBUG_NORMAL, "EP0_STALL ... what now?\n");
-+ dev->ep0state = EP0_IDLE;
-+ break;
-+ }
-+}
-+
-+/*
-+ * handle_ep - Manage I/O endpoints
-+ */
-+
-+static void s3c2410_udc_handle_ep(struct s3c2410_ep *ep)
-+{
-+ struct s3c2410_request *req;
-+ int is_in = ep->bEndpointAddress & USB_DIR_IN;
-+ u32 ep_csr1;
-+ u32 idx;
-+
-+ if (likely (!list_empty(&ep->queue)))
-+ req = list_entry(ep->queue.next,
-+ struct s3c2410_request, queue);
-+ else
-+ req = NULL;
-+
-+ idx = ep->bEndpointAddress & 0x7F;
-+
-+ if (is_in) {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr1 = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ dprintk(DEBUG_VERBOSE, "ep%01d write csr:%02x %d\n",
-+ idx, ep_csr1, req ? 1 : 0);
-+
-+ if (ep_csr1 & S3C2410_UDC_ICSR1_SENTSTL) {
-+ dprintk(DEBUG_VERBOSE, "st\n");
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr1 & ~S3C2410_UDC_ICSR1_SENTSTL,
-+ S3C2410_UDC_IN_CSR1_REG);
-+ return;
-+ }
-+
-+ if (!(ep_csr1 & S3C2410_UDC_ICSR1_PKTRDY) && req) {
-+ s3c2410_udc_write_fifo(ep,req);
-+ }
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr1 = udc_read(S3C2410_UDC_OUT_CSR1_REG);
-+ dprintk(DEBUG_VERBOSE, "ep%01d rd csr:%02x\n", idx, ep_csr1);
-+
-+ if (ep_csr1 & S3C2410_UDC_OCSR1_SENTSTL) {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ udc_write(ep_csr1 & ~S3C2410_UDC_OCSR1_SENTSTL,
-+ S3C2410_UDC_OUT_CSR1_REG);
-+ return;
-+ }
-+
-+ if ((ep_csr1 & S3C2410_UDC_OCSR1_PKTRDY) && req) {
-+ s3c2410_udc_read_fifo(ep,req);
-+ }
-+ }
-+}
-+
-+#include <asm/arch/regs-irq.h>
-+
-+/*
-+ * s3c2410_udc_irq - interrupt handler
-+ */
-+static irqreturn_t s3c2410_udc_irq(int irq, void *_dev)
-+{
-+ struct s3c2410_udc *dev = _dev;
-+ int usb_status;
-+ int usbd_status;
-+ int pwr_reg;
-+ int ep0csr;
-+ int i;
-+ u32 idx;
-+ unsigned long flags;
-+
-+ spin_lock_irqsave(&dev->lock, flags);
-+
-+ /* Driver connected ? */
-+ if (!dev->driver) {
-+ /* Clear interrupts */
-+ udc_write(udc_read(S3C2410_UDC_USB_INT_REG),
-+ S3C2410_UDC_USB_INT_REG);
-+ udc_write(udc_read(S3C2410_UDC_EP_INT_REG),
-+ S3C2410_UDC_EP_INT_REG);
-+ }
-+
-+ /* Save index */
-+ idx = udc_read(S3C2410_UDC_INDEX_REG);
-+
-+ /* Read status registers */
-+ usb_status = udc_read(S3C2410_UDC_USB_INT_REG);
-+ usbd_status = udc_read(S3C2410_UDC_EP_INT_REG);
-+ pwr_reg = udc_read(S3C2410_UDC_PWR_REG);
-+
-+ udc_writeb(base_addr, S3C2410_UDC_INDEX_EP0, S3C2410_UDC_INDEX_REG);
-+ ep0csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+
-+ dprintk(DEBUG_NORMAL, "usbs=%02x, usbds=%02x, pwr=%02x ep0csr=%02x\n",
-+ usb_status, usbd_status, pwr_reg, ep0csr);
-+
-+ /*
-+ * Now, handle interrupts. There's two types :
-+ * - Reset, Resume, Suspend coming -> usb_int_reg
-+ * - EP -> ep_int_reg
-+ */
-+
-+ /* RESET */
-+ if (usb_status & S3C2410_UDC_USBINT_RESET) {
-+ /* two kind of reset :
-+ * - reset start -> pwr reg = 8
-+ * - reset end -> pwr reg = 0
-+ **/
-+ dprintk(DEBUG_NORMAL, "USB reset csr %x pwr %x\n",
-+ ep0csr, pwr_reg);
-+
-+ dev->gadget.speed = USB_SPEED_UNKNOWN;
-+ udc_write(0x00, S3C2410_UDC_INDEX_REG);
-+ udc_write((dev->ep[0].ep.maxpacket & 0x7ff) >> 3,
-+ S3C2410_UDC_MAXP_REG);
-+ dev->address = 0;
-+
-+ dev->ep0state = EP0_IDLE;
-+ dev->gadget.speed = USB_SPEED_FULL;
-+
-+ /* clear interrupt */
-+ udc_write(S3C2410_UDC_USBINT_RESET,
-+ S3C2410_UDC_USB_INT_REG);
-+
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ spin_unlock_irqrestore(&dev->lock, flags);
-+ return IRQ_HANDLED;
-+ }
-+
-+ /* RESUME */
-+ if (usb_status & S3C2410_UDC_USBINT_RESUME) {
-+ dprintk(DEBUG_NORMAL, "USB resume\n");
-+
-+ /* clear interrupt */
-+ udc_write(S3C2410_UDC_USBINT_RESUME,
-+ S3C2410_UDC_USB_INT_REG);
-+
-+ if (dev->gadget.speed != USB_SPEED_UNKNOWN
-+ && dev->driver
-+ && dev->driver->resume)
-+ dev->driver->resume(&dev->gadget);
-+ }
-+
-+ /* SUSPEND */
-+ if (usb_status & S3C2410_UDC_USBINT_SUSPEND) {
-+ dprintk(DEBUG_NORMAL, "USB suspend\n");
-+
-+ /* clear interrupt */
-+ udc_write(S3C2410_UDC_USBINT_SUSPEND,
-+ S3C2410_UDC_USB_INT_REG);
-+
-+ if (dev->gadget.speed != USB_SPEED_UNKNOWN
-+ && dev->driver
-+ && dev->driver->suspend)
-+ dev->driver->suspend(&dev->gadget);
-+
-+ dev->ep0state = EP0_IDLE;
-+ }
-+
-+ /* EP */
-+ /* control traffic */
-+ /* check on ep0csr != 0 is not a good idea as clearing in_pkt_ready
-+ * generate an interrupt
-+ */
-+ if (usbd_status & S3C2410_UDC_INT_EP0) {
-+ dprintk(DEBUG_VERBOSE, "USB ep0 irq\n");
-+ /* Clear the interrupt bit by setting it to 1 */
-+ udc_write(S3C2410_UDC_INT_EP0, S3C2410_UDC_EP_INT_REG);
-+ s3c2410_udc_handle_ep0(dev);
-+ }
-+
-+ /* endpoint data transfers */
-+ for (i = 1; i < S3C2410_ENDPOINTS; i++) {
-+ u32 tmp = 1 << i;
-+ if (usbd_status & tmp) {
-+ dprintk(DEBUG_VERBOSE, "USB ep%d irq\n", i);
-+
-+ /* Clear the interrupt bit by setting it to 1 */
-+ udc_write(tmp, S3C2410_UDC_EP_INT_REG);
-+ s3c2410_udc_handle_ep(&dev->ep[i]);
-+ }
-+ }
-+
-+ dprintk(DEBUG_VERBOSE, "irq: %d s3c2410_udc_done.\n", irq);
-+
-+ /* Restore old index */
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+
-+ spin_unlock_irqrestore(&dev->lock, flags);
-+
-+ return IRQ_HANDLED;
-+}
-+/*------------------------- s3c2410_ep_ops ----------------------------------*/
-+
-+static inline struct s3c2410_ep *to_s3c2410_ep(struct usb_ep *ep)
-+{
-+ return container_of(ep, struct s3c2410_ep, ep);
-+}
-+
-+static inline struct s3c2410_udc *to_s3c2410_udc(struct usb_gadget *gadget)
-+{
-+ return container_of(gadget, struct s3c2410_udc, gadget);
-+}
-+
-+static inline struct s3c2410_request *to_s3c2410_req(struct usb_request *req)
-+{
-+ return container_of(req, struct s3c2410_request, req);
-+}
-+
-+/*
-+ * s3c2410_udc_ep_enable
-+ */
-+static int s3c2410_udc_ep_enable(struct usb_ep *_ep,
-+ const struct usb_endpoint_descriptor *desc)
-+{
-+ struct s3c2410_udc *dev;
-+ struct s3c2410_ep *ep;
-+ u32 max, tmp;
-+ unsigned long flags;
-+ u32 csr1,csr2;
-+ u32 int_en_reg;
-+
-+ ep = to_s3c2410_ep(_ep);
-+
-+ if (!_ep || !desc || ep->desc
-+ || _ep->name == ep0name
-+ || desc->bDescriptorType != USB_DT_ENDPOINT)
-+ return -EINVAL;
-+
-+ dev = ep->dev;
-+ if (!dev->driver || dev->gadget.speed == USB_SPEED_UNKNOWN)
-+ return -ESHUTDOWN;
-+
-+ max = le16_to_cpu(desc->wMaxPacketSize) & 0x1fff;
-+
-+ local_irq_save (flags);
-+ _ep->maxpacket = max & 0x7ff;
-+ ep->desc = desc;
-+ ep->halted = 0;
-+ ep->bEndpointAddress = desc->bEndpointAddress;
-+
-+ /* set max packet */
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(max >> 3, S3C2410_UDC_MAXP_REG);
-+
-+ /* set type, direction, address; reset fifo counters */
-+ if (desc->bEndpointAddress & USB_DIR_IN) {
-+ csr1 = S3C2410_UDC_ICSR1_FFLUSH|S3C2410_UDC_ICSR1_CLRDT;
-+ csr2 = S3C2410_UDC_ICSR2_MODEIN|S3C2410_UDC_ICSR2_DMAIEN;
-+
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr1, S3C2410_UDC_IN_CSR1_REG);
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr2, S3C2410_UDC_IN_CSR2_REG);
-+ } else {
-+ /* don't flush in fifo or it will cause endpoint interrupt */
-+ csr1 = S3C2410_UDC_ICSR1_CLRDT;
-+ csr2 = S3C2410_UDC_ICSR2_DMAIEN;
-+
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr1, S3C2410_UDC_IN_CSR1_REG);
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr2, S3C2410_UDC_IN_CSR2_REG);
-+
-+ csr1 = S3C2410_UDC_OCSR1_FFLUSH | S3C2410_UDC_OCSR1_CLRDT;
-+ csr2 = S3C2410_UDC_OCSR2_DMAIEN;
-+
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr1, S3C2410_UDC_OUT_CSR1_REG);
-+ udc_write(ep->num, S3C2410_UDC_INDEX_REG);
-+ udc_write(csr2, S3C2410_UDC_OUT_CSR2_REG);
-+ }
-+
-+ /* enable irqs */
-+ int_en_reg = udc_read(S3C2410_UDC_EP_INT_EN_REG);
-+ udc_write(int_en_reg | (1 << ep->num), S3C2410_UDC_EP_INT_EN_REG);
-+
-+ /* print some debug message */
-+ tmp = desc->bEndpointAddress;
-+ dprintk (DEBUG_NORMAL, "enable %s(%d) ep%x%s-blk max %02x\n",
-+ _ep->name,ep->num, tmp,
-+ desc->bEndpointAddress & USB_DIR_IN ? "in" : "out", max);
-+
-+ local_irq_restore (flags);
-+ s3c2410_udc_set_halt(_ep, 0);
-+
-+ return 0;
-+}
-+
-+/*
-+ * s3c2410_udc_ep_disable
-+ */
-+static int s3c2410_udc_ep_disable(struct usb_ep *_ep)
-+{
-+ struct s3c2410_ep *ep = to_s3c2410_ep(_ep);
-+ unsigned long flags;
-+ u32 int_en_reg;
-+
-+ if (!_ep || !ep->desc) {
-+ dprintk(DEBUG_NORMAL, "%s not enabled\n",
-+ _ep ? ep->ep.name : NULL);
-+ return -EINVAL;
-+ }
-+
-+ local_irq_save(flags);
-+
-+ dprintk(DEBUG_NORMAL, "ep_disable: %s\n", _ep->name);
-+
-+ ep->desc = NULL;
-+ ep->halted = 1;
-+
-+ s3c2410_udc_nuke (ep->dev, ep, -ESHUTDOWN);
-+
-+ /* disable irqs */
-+ int_en_reg = udc_read(S3C2410_UDC_EP_INT_EN_REG);
-+ udc_write(int_en_reg & ~(1<<ep->num), S3C2410_UDC_EP_INT_EN_REG);
-+
-+ local_irq_restore(flags);
-+
-+ dprintk(DEBUG_NORMAL, "%s disabled\n", _ep->name);
-+
-+ return 0;
-+}
-+
-+/*
-+ * s3c2410_udc_alloc_request
-+ */
-+static struct usb_request *
-+s3c2410_udc_alloc_request(struct usb_ep *_ep, gfp_t mem_flags)
-+{
-+ struct s3c2410_request *req;
-+
-+ dprintk(DEBUG_VERBOSE,"%s(%p,%d)\n", __func__, _ep, mem_flags);
-+
-+ if (!_ep)
-+ return NULL;
-+
-+ req = kzalloc (sizeof(struct s3c2410_request), mem_flags);
-+ if (!req)
-+ return NULL;
-+
-+ INIT_LIST_HEAD (&req->queue);
-+ return &req->req;
-+}
-+
-+/*
-+ * s3c2410_udc_free_request
-+ */
-+static void
-+s3c2410_udc_free_request(struct usb_ep *_ep, struct usb_request *_req)
-+{
-+ struct s3c2410_ep *ep = to_s3c2410_ep(_ep);
-+ struct s3c2410_request *req = to_s3c2410_req(_req);
-+
-+ dprintk(DEBUG_VERBOSE, "%s(%p,%p)\n", __func__, _ep, _req);
-+
-+ if (!ep || !_req || (!ep->desc && _ep->name != ep0name))
-+ return;
-+
-+ WARN_ON (!list_empty (&req->queue));
-+ kfree(req);
-+}
-+
-+/*
-+ * s3c2410_udc_alloc_buffer
-+ */
-+static void *s3c2410_udc_alloc_buffer(struct usb_ep *_ep,
-+ unsigned bytes, dma_addr_t *dma, gfp_t mem_flags)
-+{
-+ char *retval;
-+
-+ dprintk(DEBUG_VERBOSE, "%s()\n", __func__);
-+
-+ if (!the_controller->driver)
-+ return NULL;
-+
-+ retval = kmalloc (bytes, mem_flags);
-+ *dma = (dma_addr_t) retval;
-+ return retval;
-+}
-+
-+/*
-+ * s3c2410_udc_free_buffer
-+ */
-+static void s3c2410_udc_free_buffer (struct usb_ep *_ep, void *buf,
-+ dma_addr_t dma, unsigned bytes)
-+{
-+ dprintk(DEBUG_VERBOSE, "%s()\n", __func__);
-+
-+ if (bytes)
-+ kfree (buf);
-+}
-+
-+/*
-+ * s3c2410_udc_queue
-+ */
-+static int s3c2410_udc_queue(struct usb_ep *_ep, struct usb_request *_req,
-+ gfp_t gfp_flags)
-+{
-+ struct s3c2410_request *req = to_s3c2410_req(_req);
-+ struct s3c2410_ep *ep = to_s3c2410_ep(_ep);
-+ struct s3c2410_udc *dev;
-+ u32 ep_csr = 0;
-+ int fifo_count = 0;
-+ unsigned long flags;
-+
-+ if (unlikely (!_ep || (!ep->desc && ep->ep.name != ep0name))) {
-+ dprintk(DEBUG_NORMAL, "%s: invalid args\n", __func__);
-+ return -EINVAL;
-+ }
-+
-+ dev = ep->dev;
-+ if (unlikely (!dev->driver
-+ || dev->gadget.speed == USB_SPEED_UNKNOWN)) {
-+ return -ESHUTDOWN;
-+ }
-+
-+ local_irq_save (flags);
-+
-+ if (unlikely(!_req || !_req->complete
-+ || !_req->buf || !list_empty(&req->queue))) {
-+ if (!_req)
-+ dprintk(DEBUG_NORMAL, "%s: 1 X X X\n", __func__);
-+ else {
-+ dprintk(DEBUG_NORMAL, "%s: 0 %01d %01d %01d\n",
-+ __func__, !_req->complete,!_req->buf,
-+ !list_empty(&req->queue));
-+ }
-+
-+ local_irq_restore(flags);
-+ return -EINVAL;
-+ }
-+
-+ _req->status = -EINPROGRESS;
-+ _req->actual = 0;
-+
-+ dprintk(DEBUG_VERBOSE, "%s: ep%x len %d\n",
-+ __func__, ep->bEndpointAddress, _req->length);
-+
-+ if (ep->bEndpointAddress) {
-+ udc_write(ep->bEndpointAddress & 0x7F, S3C2410_UDC_INDEX_REG);
-+
-+ ep_csr = udc_read((ep->bEndpointAddress & USB_DIR_IN)
-+ ? S3C2410_UDC_IN_CSR1_REG
-+ : S3C2410_UDC_OUT_CSR1_REG);
-+ fifo_count = s3c2410_udc_fifo_count_out();
-+ } else {
-+ udc_write(0, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read(S3C2410_UDC_IN_CSR1_REG);
-+ fifo_count = s3c2410_udc_fifo_count_out();
-+ }
-+
-+ /* kickstart this i/o queue? */
-+ if (list_empty(&ep->queue) && !ep->halted) {
-+ if (ep->bEndpointAddress == 0 /* ep0 */) {
-+ switch (dev->ep0state) {
-+ case EP0_IN_DATA_PHASE:
-+ if (!(ep_csr&S3C2410_UDC_EP0_CSR_IPKRDY)
-+ && s3c2410_udc_write_fifo(ep,
-+ req)) {
-+ dev->ep0state = EP0_IDLE;
-+ req = NULL;
-+ }
-+ break;
-+
-+ case EP0_OUT_DATA_PHASE:
-+ if ((!_req->length)
-+ || ((ep_csr & S3C2410_UDC_OCSR1_PKTRDY)
-+ && s3c2410_udc_read_fifo(ep,
-+ req))) {
-+ dev->ep0state = EP0_IDLE;
-+ req = NULL;
-+ }
-+ break;
-+
-+ default:
-+ local_irq_restore(flags);
-+ return -EL2HLT;
-+ }
-+ } else if ((ep->bEndpointAddress & USB_DIR_IN) != 0
-+ && (!(ep_csr&S3C2410_UDC_OCSR1_PKTRDY))
-+ && s3c2410_udc_write_fifo(ep, req)) {
-+ req = NULL;
-+ } else if ((ep_csr & S3C2410_UDC_OCSR1_PKTRDY)
-+ && fifo_count
-+ && s3c2410_udc_read_fifo(ep, req)) {
-+ req = NULL;
-+ }
-+ }
-+
-+ /* pio or dma irq handler advances the queue. */
-+ if (likely (req != 0))
-+ list_add_tail(&req->queue, &ep->queue);
-+
-+ local_irq_restore(flags);
-+
-+ dprintk(DEBUG_VERBOSE, "%s ok\n", __func__);
-+ return 0;
-+}
-+
-+/*
-+ * s3c2410_udc_dequeue
-+ */
-+static int s3c2410_udc_dequeue(struct usb_ep *_ep, struct usb_request *_req)
-+{
-+ struct s3c2410_ep *ep = to_s3c2410_ep(_ep);
-+ struct s3c2410_udc *udc;
-+ int retval = -EINVAL;
-+ unsigned long flags;
-+ struct s3c2410_request *req = NULL;
-+
-+ dprintk(DEBUG_VERBOSE, "%s(%p,%p)\n", __func__, _ep, _req);
-+
-+ if (!the_controller->driver)
-+ return -ESHUTDOWN;
-+
-+ if (!_ep || !_req)
-+ return retval;
-+
-+ udc = to_s3c2410_udc(ep->gadget);
-+
-+ local_irq_save (flags);
-+
-+ list_for_each_entry (req, &ep->queue, queue) {
-+ if (&req->req == _req) {
-+ list_del_init (&req->queue);
-+ _req->status = -ECONNRESET;
-+ retval = 0;
-+ break;
-+ }
-+ }
-+
-+ if (retval == 0) {
-+ dprintk(DEBUG_VERBOSE,
-+ "dequeued req %p from %s, len %d buf %p\n",
-+ req, _ep->name, _req->length, _req->buf);
-+
-+ s3c2410_udc_done(ep, req, -ECONNRESET);
-+ }
-+
-+ local_irq_restore (flags);
-+ return retval;
-+}
-+
-+/*
-+ * s3c2410_udc_set_halt
-+ */
-+static int s3c2410_udc_set_halt(struct usb_ep *_ep, int value)
-+{
-+ struct s3c2410_ep *ep = to_s3c2410_ep(_ep);
-+ u32 ep_csr = 0;
-+ unsigned long flags;
-+ u32 idx;
-+
-+ if (unlikely (!_ep || (!ep->desc && ep->ep.name != ep0name))) {
-+ dprintk(DEBUG_NORMAL, "%s: inval 2\n", __func__);
-+ return -EINVAL;
-+ }
-+
-+ local_irq_save (flags);
-+
-+ idx = ep->bEndpointAddress & 0x7F;
-+
-+ if (idx == 0) {
-+ s3c2410_udc_set_ep0_ss(base_addr);
-+ s3c2410_udc_set_ep0_de_out(base_addr);
-+ } else {
-+ udc_write(idx, S3C2410_UDC_INDEX_REG);
-+ ep_csr = udc_read((ep->bEndpointAddress &USB_DIR_IN)
-+ ? S3C2410_UDC_IN_CSR1_REG
-+ : S3C2410_UDC_OUT_CSR1_REG);
-+
-+ if ((ep->bEndpointAddress & USB_DIR_IN) != 0) {
-+ if (value)
-+ udc_write(ep_csr | S3C2410_UDC_ICSR1_SENDSTL,
-+ S3C2410_UDC_IN_CSR1_REG);
-+ else {
-+ ep_csr &= ~S3C2410_UDC_ICSR1_SENDSTL;
-+ udc_write(ep_csr, S3C2410_UDC_IN_CSR1_REG);
-+ ep_csr |= S3C2410_UDC_ICSR1_CLRDT;
-+ udc_write(ep_csr, S3C2410_UDC_IN_CSR1_REG);
-+ }
-+ } else {
-+ if (value)
-+ udc_write(ep_csr | S3C2410_UDC_OCSR1_SENDSTL,
-+ S3C2410_UDC_OUT_CSR1_REG);
-+ else {
-+ ep_csr &= ~S3C2410_UDC_OCSR1_SENDSTL;
-+ udc_write(ep_csr, S3C2410_UDC_OUT_CSR1_REG);
-+ ep_csr |= S3C2410_UDC_OCSR1_CLRDT;
-+ udc_write(ep_csr, S3C2410_UDC_OUT_CSR1_REG);
-+ }
-+ }
-+ }
-+
-+ ep->halted = value ? 1 : 0;
-+ local_irq_restore (flags);
-+
-+ return 0;
-+}
-+
-+static const struct usb_ep_ops s3c2410_ep_ops = {
-+ .enable = s3c2410_udc_ep_enable,
-+ .disable = s3c2410_udc_ep_disable,
-+
-+ .alloc_request = s3c2410_udc_alloc_request,
-+ .free_request = s3c2410_udc_free_request,
-+
-+ .alloc_buffer = s3c2410_udc_alloc_buffer,
-+ .free_buffer = s3c2410_udc_free_buffer,
-+
-+ .queue = s3c2410_udc_queue,
-+ .dequeue = s3c2410_udc_dequeue,
-+
-+ .set_halt = s3c2410_udc_set_halt,
-+};
-+
-+/*------------------------- usb_gadget_ops ----------------------------------*/
-+
-+/*
-+ * s3c2410_udc_get_frame
-+ */
-+static int s3c2410_udc_get_frame(struct usb_gadget *_gadget)
-+{
-+ int tmp;
-+
-+ dprintk(DEBUG_VERBOSE, "%s()\n", __func__);
-+
-+ tmp = udc_read(S3C2410_UDC_FRAME_NUM2_REG) << 8;
-+ tmp |= udc_read(S3C2410_UDC_FRAME_NUM1_REG);
-+ return tmp;
-+}
-+
-+/*
-+ * s3c2410_udc_wakeup
-+ */
-+static int s3c2410_udc_wakeup(struct usb_gadget *_gadget)
-+{
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+ return 0;
-+}
-+
-+/*
-+ * s3c2410_udc_set_selfpowered
-+ */
-+static int s3c2410_udc_set_selfpowered(struct usb_gadget *gadget, int value)
-+{
-+ struct s3c2410_udc *udc = to_s3c2410_udc(gadget);
-+
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ if (value)
-+ udc->devstatus |= (1 << USB_DEVICE_SELF_POWERED);
-+ else
-+ udc->devstatus &= ~(1 << USB_DEVICE_SELF_POWERED);
-+
-+ return 0;
-+}
-+
-+static void s3c2410_udc_disable(struct s3c2410_udc *dev);
-+static void s3c2410_udc_enable(struct s3c2410_udc *dev);
-+
-+static int s3c2410_udc_set_pullup(struct s3c2410_udc *udc, int is_on)
-+{
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ if (udc_info && udc_info->udc_command) {
-+ if (is_on)
-+ s3c2410_udc_enable(udc);
-+ else {
-+ if (udc->gadget.speed != USB_SPEED_UNKNOWN) {
-+ if (udc->driver && udc->driver->disconnect)
-+ udc->driver->disconnect(&udc->gadget);
-+
-+ }
-+ s3c2410_udc_disable(udc);
-+ }
-+ }
-+ else
-+ return -EOPNOTSUPP;
-+
-+ return 0;
-+}
-+
-+static int s3c2410_udc_vbus_session(struct usb_gadget *gadget, int is_active)
-+{
-+ struct s3c2410_udc *udc = to_s3c2410_udc(gadget);
-+
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ udc->vbus = (is_active != 0);
-+ s3c2410_udc_set_pullup(udc, is_active);
-+ return 0;
-+}
-+
-+static int s3c2410_udc_pullup(struct usb_gadget *gadget, int is_on)
-+{
-+ struct s3c2410_udc *udc = to_s3c2410_udc(gadget);
-+
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ s3c2410_udc_set_pullup(udc, is_on ? 0 : 1);
-+ return 0;
-+}
-+
-+static irqreturn_t s3c2410_udc_vbus_irq(int irq, void *_dev)
-+{
-+ struct s3c2410_udc *dev = _dev;
-+ unsigned int value;
-+
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+ value = s3c2410_gpio_getpin(udc_info->vbus_pin);
-+
-+ if (udc_info->vbus_pin_inverted)
-+ value = !value;
-+
-+ if (value != dev->vbus)
-+ s3c2410_udc_vbus_session(&dev->gadget, value);
-+
-+ return IRQ_HANDLED;
-+}
-+
-+static int s3c2410_vbus_draw(struct usb_gadget *_gadget, unsigned ma)
-+{
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ if (udc_info && udc_info->vbus_draw) {
-+ udc_info->vbus_draw(ma);
-+ return 0;
-+ }
-+
-+ return -ENOTSUPP;
-+}
-+
-+static const struct usb_gadget_ops s3c2410_ops = {
-+ .get_frame = s3c2410_udc_get_frame,
-+ .wakeup = s3c2410_udc_wakeup,
-+ .set_selfpowered = s3c2410_udc_set_selfpowered,
-+ .pullup = s3c2410_udc_pullup,
-+ .vbus_session = s3c2410_udc_vbus_session,
-+ .vbus_draw = s3c2410_vbus_draw,
-+};
-+
-+/*------------------------- gadget driver handling---------------------------*/
-+/*
-+ * s3c2410_udc_disable
-+ */
-+static void s3c2410_udc_disable(struct s3c2410_udc *dev)
-+{
-+ dprintk(DEBUG_NORMAL, "%s()\n", __func__);
-+
-+ /* Disable all interrupts */
-+ udc_write(0x00, S3C2410_UDC_USB_INT_EN_REG);
-+ udc_write(0x00, S3C2410_UDC_EP_INT_EN_REG);
-+
-+ /* Clear the interrupt registers */
-+ udc_write(S3C2410_UDC_USBINT_RESET
-+ | S3C2410_UDC_USBINT_RESUME
-+ | S3C2410_UDC_USBINT_SUSPEND,
-+ S3C2410_UDC_USB_INT_REG);
-+
-+ udc_write(0x1F, S3C2410_UDC_EP_INT_REG);
-+
-+ /* Good bye, cruel world */
-+ if (udc_info && udc_info->udc_command)
-+ udc_info->udc_command(S3C2410_UDC_P_DISABLE);
-+
-+ /* Set speed to unknown */
-+ dev->gadget.speed = USB_SPEED_UNKNOWN;
-+}
-+
-+/*
-+ * s3c2410_udc_reinit
-+ */
-+static void s3c2410_udc_reinit(struct s3c2410_udc *dev)
-+{
-+ u32 i;
-+
-+ /* device/ep0 records init */
-+ INIT_LIST_HEAD (&dev->gadget.ep_list);
-+ INIT_LIST_HEAD (&dev->gadget.ep0->ep_list);
-+ dev->ep0state = EP0_IDLE;
-+
-+ for (i = 0; i < S3C2410_ENDPOINTS; i++) {
-+ struct s3c2410_ep *ep = &dev->ep[i];
-+
-+ if (i != 0)
-+ list_add_tail (&ep->ep.ep_list, &dev->gadget.ep_list);
-+
-+ ep->dev = dev;
-+ ep->desc = NULL;
-+ ep->halted = 0;
-+ INIT_LIST_HEAD (&ep->queue);
-+ }
-+}
-+
-+/*
-+ * s3c2410_udc_enable
-+ */
-+static void s3c2410_udc_enable(struct s3c2410_udc *dev)
-+{
-+ int i;
-+
-+ dprintk(DEBUG_NORMAL, "s3c2410_udc_enable called\n");
-+
-+ /* dev->gadget.speed = USB_SPEED_UNKNOWN; */
-+ dev->gadget.speed = USB_SPEED_FULL;
-+
-+ /* Set MAXP for all endpoints */
-+ for (i = 0; i < S3C2410_ENDPOINTS; i++) {
-+ udc_write(i, S3C2410_UDC_INDEX_REG);
-+ udc_write((dev->ep[i].ep.maxpacket & 0x7ff) >> 3,
-+ S3C2410_UDC_MAXP_REG);
-+ }
-+
-+ /* Set default power state */
-+ udc_write(DEFAULT_POWER_STATE, S3C2410_UDC_PWR_REG);
-+
-+ /* Enable reset and suspend interrupt interrupts */
-+ udc_write(S3C2410_UDC_USBINT_RESET | S3C2410_UDC_USBINT_SUSPEND,
-+ S3C2410_UDC_USB_INT_EN_REG);
-+
-+ /* Enable ep0 interrupt */
-+ udc_write(S3C2410_UDC_INT_EP0, S3C2410_UDC_EP_INT_EN_REG);
-+
-+ /* time to say "hello, world" */
-+ if (udc_info && udc_info->udc_command)
-+ udc_info->udc_command(S3C2410_UDC_P_ENABLE);
-+}
-+
-+/*
-+ * usb_gadget_register_driver
-+ */
-+int usb_gadget_register_driver(struct usb_gadget_driver *driver)
-+{
-+ struct s3c2410_udc *udc = the_controller;
-+ int retval;
-+
-+ dprintk(DEBUG_NORMAL, "usb_gadget_register_driver() '%s'\n",
-+ driver->driver.name);
-+
-+ /* Sanity checks */
-+ if (!udc)
-+ return -ENODEV;
-+
-+ if (udc->driver)
-+ return -EBUSY;
-+
-+ if (!driver->bind || !driver->setup
-+ || driver->speed != USB_SPEED_FULL) {
-+ printk(KERN_ERR "Invalid driver: bind %p setup %p speed %d\n",
-+ driver->bind, driver->setup, driver->speed);
-+ return -EINVAL;
-+ }
-+#if defined(MODULE)
-+ if (!driver->unbind) {
-+ printk(KERN_ERR "Invalid driver: no unbind method\n");
-+ return -EINVAL;
-+ }
-+#endif
-+
-+ /* Hook the driver */
-+ udc->driver = driver;
-+ udc->gadget.dev.driver = &driver->driver;
-+
-+ /* Bind the driver */
-+ if ((retval = device_add(&udc->gadget.dev)) != 0) {
-+ printk(KERN_ERR "Error in device_add() : %d\n",retval);
-+ goto register_error;
-+ }
-+
-+ dprintk(DEBUG_NORMAL, "binding gadget driver '%s'\n",
-+ driver->driver.name);
-+
-+ if ((retval = driver->bind (&udc->gadget)) != 0) {
-+ device_del(&udc->gadget.dev);
-+ goto register_error;
-+ }
-+
-+ /* Enable udc */
-+ s3c2410_udc_enable(udc);
-+
-+ return 0;
-+
-+register_error:
-+ udc->driver = NULL;
-+ udc->gadget.dev.driver = NULL;
-+ return retval;
-+}
-+
-+/*
-+ * usb_gadget_unregister_driver
-+ */
-+int usb_gadget_unregister_driver(struct usb_gadget_driver *driver)
-+{
-+ struct s3c2410_udc *udc = the_controller;
-+
-+ if (!udc)
-+ return -ENODEV;
-+
-+ if (!driver || driver != udc->driver || !driver->unbind)
-+ return -EINVAL;
-+
-+ dprintk(DEBUG_NORMAL,"usb_gadget_register_driver() '%s'\n",
-+ driver->driver.name);
-+
-+ if (driver->disconnect)
-+ driver->disconnect(&udc->gadget);
-+
-+ device_del(&udc->gadget.dev);
-+ udc->driver = NULL;
-+
-+ /* Disable udc */
-+ s3c2410_udc_disable(udc);
-+
-+ return 0;
-+}
-+
-+/*---------------------------------------------------------------------------*/
-+static struct s3c2410_udc memory = {
-+ .gadget = {
-+ .ops = &s3c2410_ops,
-+ .ep0 = &memory.ep[0].ep,
-+ .name = gadget_name,
-+ .dev = {
-+ .bus_id = "gadget",
-+ },
-+ },
-+
-+ /* control endpoint */
-+ .ep[0] = {
-+ .num = 0,
-+ .ep = {
-+ .name = ep0name,
-+ .ops = &s3c2410_ep_ops,
-+ .maxpacket = EP0_FIFO_SIZE,
-+ },
-+ .dev = &memory,
-+ },
-+
-+ /* first group of endpoints */
-+ .ep[1] = {
-+ .num = 1,
-+ .ep = {
-+ .name = "ep1-bulk",
-+ .ops = &s3c2410_ep_ops,
-+ .maxpacket = EP_FIFO_SIZE,
-+ },
-+ .dev = &memory,
-+ .fifo_size = EP_FIFO_SIZE,
-+ .bEndpointAddress = 1,
-+ .bmAttributes = USB_ENDPOINT_XFER_BULK,
-+ },
-+ .ep[2] = {
-+ .num = 2,
-+ .ep = {
-+ .name = "ep2-bulk",
-+ .ops = &s3c2410_ep_ops,
-+ .maxpacket = EP_FIFO_SIZE,
-+ },
-+ .dev = &memory,
-+ .fifo_size = EP_FIFO_SIZE,
-+ .bEndpointAddress = 2,
-+ .bmAttributes = USB_ENDPOINT_XFER_BULK,
-+ },
-+ .ep[3] = {
-+ .num = 3,
-+ .ep = {
-+ .name = "ep3-bulk",
-+ .ops = &s3c2410_ep_ops,
-+ .maxpacket = EP_FIFO_SIZE,
-+ },
-+ .dev = &memory,
-+ .fifo_size = EP_FIFO_SIZE,
-+ .bEndpointAddress = 3,
-+ .bmAttributes = USB_ENDPOINT_XFER_BULK,
-+ },
-+ .ep[4] = {
-+ .num = 4,
-+ .ep = {
-+ .name = "ep4-bulk",
-+ .ops = &s3c2410_ep_ops,
-+ .maxpacket = EP_FIFO_SIZE,
-+ },
-+ .dev = &memory,
-+ .fifo_size = EP_FIFO_SIZE,
-+ .bEndpointAddress = 4,
-+ .bmAttributes = USB_ENDPOINT_XFER_BULK,
-+ }
-+
-+};
-+
-+/*
-+ * probe - binds to the platform device
-+ */
-+static int s3c2410_udc_probe(struct platform_device *pdev)
-+{
-+ struct s3c2410_udc *udc = &memory;
-+ struct device *dev = &pdev->dev;
-+ int retval;
-+ unsigned int irq;
-+
-+ dev_dbg(dev, "%s()\n", __func__);
-+
-+ usb_bus_clock = clk_get(NULL, "usb-bus-gadget");
-+ if (IS_ERR(usb_bus_clock)) {
-+ dev_err(dev, "failed to get usb bus clock source\n");
-+ return PTR_ERR(usb_bus_clock);
-+ }
-+
-+ clk_enable(usb_bus_clock);
-+
-+ udc_clock = clk_get(NULL, "usb-device");
-+ if (IS_ERR(udc_clock)) {
-+ dev_err(dev, "failed to get udc clock source\n");
-+ return PTR_ERR(udc_clock);
-+ }
-+
-+ clk_enable(udc_clock);
-+
-+ mdelay(10);
-+
-+ dev_dbg(dev, "got and enabled clocks\n");
-+
-+ if (strncmp(pdev->name, "s3c2440", 7) == 0) {
-+ dev_info(dev, "S3C2440: increasing FIFO to 128 bytes\n");
-+ memory.ep[1].fifo_size = S3C2440_EP_FIFO_SIZE;
-+ memory.ep[2].fifo_size = S3C2440_EP_FIFO_SIZE;
-+ memory.ep[3].fifo_size = S3C2440_EP_FIFO_SIZE;
-+ memory.ep[4].fifo_size = S3C2440_EP_FIFO_SIZE;
-+ }
-+
-+ spin_lock_init (&udc->lock);
-+ udc_info = pdev->dev.platform_data;
-+
-+ rsrc_start = S3C2410_PA_USBDEV;
-+ rsrc_len = S3C24XX_SZ_USBDEV;
-+
-+ if (!request_mem_region(rsrc_start, rsrc_len, gadget_name))
-+ return -EBUSY;
-+
-+ base_addr = ioremap(rsrc_start, rsrc_len);
-+ if (!base_addr) {
-+ retval = -ENOMEM;
-+ goto err_mem;
-+ }
-+
-+ device_initialize(&udc->gadget.dev);
-+ udc->gadget.dev.parent = &pdev->dev;
-+ udc->gadget.dev.dma_mask = pdev->dev.dma_mask;
-+
-+ the_controller = udc;
-+ platform_set_drvdata(pdev, udc);
-+
-+ s3c2410_udc_disable(udc);
-+ s3c2410_udc_reinit(udc);
-+
-+ /* irq setup after old hardware state is cleaned up */
-+ retval = request_irq(IRQ_USBD, s3c2410_udc_irq,
-+ IRQF_DISABLED, gadget_name, udc);
-+
-+ if (retval != 0) {
-+ dev_err(dev, "cannot get irq %i, err %d\n", IRQ_USBD, retval);
-+ retval = -EBUSY;
-+ goto err_map;
-+ }
-+
-+ dev_dbg(dev, "got irq %i\n", IRQ_USBD);
-+
-+ if (udc_info && udc_info->vbus_pin > 0) {
-+ irq = s3c2410_gpio_getirq(udc_info->vbus_pin);
-+ retval = request_irq(irq, s3c2410_udc_vbus_irq,
-+ IRQF_DISABLED | IRQF_TRIGGER_RISING
-+ | IRQF_TRIGGER_FALLING,
-+ gadget_name, udc);
-+
-+ if (retval != 0) {
-+ dev_err(dev, "can't get vbus irq %i, err %d\n",
-+ irq, retval);
-+ retval = -EBUSY;
-+ goto err_int;
-+ }
-+
-+ dev_dbg(dev, "got irq %i\n", irq);
-+ } else {
-+ udc->vbus = 1;
-+ }
-+
-+ if (s3c2410_udc_debugfs_root) {
-+ udc->regs_info = debugfs_create_file("registers", S_IRUGO,
-+ s3c2410_udc_debugfs_root,
-+ udc, &s3c2410_udc_debugfs_fops);
-+ if (IS_ERR(udc->regs_info)) {
-+ dev_warn(dev, "debugfs file creation failed %ld\n",
-+ PTR_ERR(udc->regs_info));
-+ udc->regs_info = NULL;
-+ }
-+ }
-+
-+ dev_dbg(dev, "probe ok\n");
-+
-+ return 0;
-+
-+err_int:
-+ free_irq(IRQ_USBD, udc);
-+err_map:
-+ iounmap(base_addr);
-+err_mem:
-+ release_mem_region(rsrc_start, rsrc_len);
-+
-+ return retval;
-+}
-+
-+/*
-+ * s3c2410_udc_remove
-+ */
-+static int s3c2410_udc_remove(struct platform_device *pdev)
-+{
-+ struct s3c2410_udc *udc = platform_get_drvdata(pdev);
-+ unsigned int irq;
-+
-+ dev_dbg(&pdev->dev, "%s()\n", __func__);
-+ if (udc->driver)
-+ return -EBUSY;
-+
-+ debugfs_remove(udc->regs_info);
-+
-+ if (udc_info && udc_info->vbus_pin > 0) {
-+ irq = s3c2410_gpio_getirq(udc_info->vbus_pin);
-+ free_irq(irq, udc);
-+ }
-+
-+ free_irq(IRQ_USBD, udc);
-+
-+ iounmap(base_addr);
-+ release_mem_region(rsrc_start, rsrc_len);
-+
-+ platform_set_drvdata(pdev, NULL);
-+
-+ if (!IS_ERR(udc_clock) && udc_clock != NULL) {
-+ clk_disable(udc_clock);
-+ clk_put(udc_clock);
-+ udc_clock = NULL;
-+ }
-+
-+ if (!IS_ERR(usb_bus_clock) && usb_bus_clock != NULL) {
-+ clk_disable(usb_bus_clock);
-+ clk_put(usb_bus_clock);
-+ usb_bus_clock = NULL;
-+ }
-+
-+ dev_dbg(&pdev->dev, "%s: remove ok\n", __func__);
-+ return 0;
-+}
-+
-+#ifdef CONFIG_PM
-+static int s3c2410_udc_suspend(struct platform_device *pdev, pm_message_t message)
-+{
-+ if (udc_info && udc_info->udc_command)
-+ udc_info->udc_command(S3C2410_UDC_P_DISABLE);
-+
-+ return 0;
-+}
-+
-+static int s3c2410_udc_resume(struct platform_device *pdev)
-+{
-+ if (udc_info && udc_info->udc_command)
-+ udc_info->udc_command(S3C2410_UDC_P_ENABLE);
-+
-+ return 0;
-+}
-+#else
-+#define s3c2410_udc_suspend NULL
-+#define s3c2410_udc_resume NULL
-+#endif
-+
-+static struct platform_driver udc_driver_2410 = {
-+ .driver = {
-+ .name = "s3c2410-usbgadget",
-+ .owner = THIS_MODULE,
-+ },
-+ .probe = s3c2410_udc_probe,
-+ .remove = s3c2410_udc_remove,
-+ .suspend = s3c2410_udc_suspend,
-+ .resume = s3c2410_udc_resume,
-+};
-+
-+static struct platform_driver udc_driver_2440 = {
-+ .driver = {
-+ .name = "s3c2440-usbgadget",
-+ .owner = THIS_MODULE,
-+ },
-+ .probe = s3c2410_udc_probe,
-+ .remove = s3c2410_udc_remove,
-+ .suspend = s3c2410_udc_suspend,
-+ .resume = s3c2410_udc_resume,
-+};
-+
-+static int __init udc_init(void)
-+{
-+ int retval;
-+
-+ dprintk(DEBUG_NORMAL, "%s: version %s\n", gadget_name, DRIVER_VERSION);
-+
-+ s3c2410_udc_debugfs_root = debugfs_create_dir(gadget_name, NULL);
-+ if (IS_ERR(s3c2410_udc_debugfs_root)) {
-+ printk(KERN_ERR "%s: debugfs dir creation failed %ld\n",
-+ gadget_name, PTR_ERR(s3c2410_udc_debugfs_root));
-+ s3c2410_udc_debugfs_root = NULL;
-+ }
-+
-+ retval = platform_driver_register(&udc_driver_2410);
-+ if (retval)
-+ goto err;
-+
-+ retval = platform_driver_register(&udc_driver_2440);
-+ if (retval)
-+ goto err;
-+
-+ return 0;
-+
-+err:
-+ debugfs_remove(s3c2410_udc_debugfs_root);
-+ return retval;
-+}
-+
-+static void __exit udc_exit(void)
-+{
-+ platform_driver_unregister(&udc_driver_2410);
-+ platform_driver_unregister(&udc_driver_2440);
-+ debugfs_remove(s3c2410_udc_debugfs_root);
-+}
-+
-+EXPORT_SYMBOL(usb_gadget_unregister_driver);
-+EXPORT_SYMBOL(usb_gadget_register_driver);
-+
-+module_init(udc_init);
-+module_exit(udc_exit);
-+
-+MODULE_AUTHOR(DRIVER_AUTHOR);
-+MODULE_DESCRIPTION(DRIVER_DESC);
-+MODULE_VERSION(DRIVER_VERSION);
-+MODULE_LICENSE("GPL");
-diff --git a/drivers/usb/gadget/s3c2410_udc.h b/drivers/usb/gadget/s3c2410_udc.h
-new file mode 100644
-index 0000000..9e0bece
---- /dev/null
-+++ b/drivers/usb/gadget/s3c2410_udc.h
-@@ -0,0 +1,110 @@
-+/*
-+ * linux/drivers/usb/gadget/s3c2410_udc.h
-+ * Samsung on-chip full speed USB device controllers
-+ *
-+ * Copyright (C) 2004-2007 Herbert Pötzl - Arnaud Patard
-+ * Additional cleanups by Ben Dooks <ben-linux at fluff.org>
-+ *
-+ * This program is free software; you can redistribute it and/or modify
-+ * it under the terms of the GNU General Public License as published by
-+ * the Free Software Foundation; either version 2 of the License, or
-+ * (at your option) any later version.
-+ *
-+ * This 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
-+ *
-+ */
-+
-+#ifndef _S3C2410_UDC_H
-+#define _S3C2410_UDC_H
-+
-+struct s3c2410_ep {
-+ struct list_head queue;
-+ unsigned long last_io; /* jiffies timestamp */
-+ struct usb_gadget *gadget;
-+ struct s3c2410_udc *dev;
-+ const struct usb_endpoint_descriptor *desc;
-+ struct usb_ep ep;
-+ u8 num;
-+
-+ unsigned short fifo_size;
-+ u8 bEndpointAddress;
-+ u8 bmAttributes;
-+
-+ unsigned halted : 1;
-+ unsigned already_seen : 1;
-+ unsigned setup_stage : 1;
-+};
-+
-+
-+/* Warning : ep0 has a fifo of 16 bytes */
-+/* Don't try to set 32 or 64 */
-+/* also testusb 14 fails wit 16 but is */
-+/* fine with 8 */
-+#define EP0_FIFO_SIZE 8
-+#define EP_FIFO_SIZE 64
-+#define DEFAULT_POWER_STATE 0x00
-+
-+#define S3C2440_EP_FIFO_SIZE 128
-+
-+static const char ep0name [] = "ep0";
-+
-+static const char *const ep_name[] = {
-+ ep0name, /* everyone has ep0 */
-+ /* s3c2410 four bidirectional bulk endpoints */
-+ "ep1-bulk", "ep2-bulk", "ep3-bulk", "ep4-bulk",
-+};
-+
-+#define S3C2410_ENDPOINTS ARRAY_SIZE(ep_name)
-+
-+struct s3c2410_request {
-+ struct list_head queue; /* ep's requests */
-+ struct usb_request req;
-+};
-+
-+enum ep0_state {
-+ EP0_IDLE,
-+ EP0_IN_DATA_PHASE,
-+ EP0_OUT_DATA_PHASE,
-+ EP0_END_XFER,
-+ EP0_STALL,
-+};
-+
-+static const char *ep0states[]= {
-+ "EP0_IDLE",
-+ "EP0_IN_DATA_PHASE",
-+ "EP0_OUT_DATA_PHASE",
-+ "EP0_END_XFER",
-+ "EP0_STALL",
-+};
-+
-+struct s3c2410_udc {
-+ spinlock_t lock;
-+
-+ struct s3c2410_ep ep[S3C2410_ENDPOINTS];
-+ int address;
-+ struct usb_gadget gadget;
-+ struct usb_gadget_driver *driver;
-+ struct s3c2410_request fifo_req;
-+ u8 fifo_buf[EP_FIFO_SIZE];
-+ u16 devstatus;
-+
-+ u32 port_status;
-+ int ep0state;
-+
-+ unsigned got_irq : 1;
-+
-+ unsigned req_std : 1;
-+ unsigned req_config : 1;
-+ unsigned req_pending : 1;
-+ u8 vbus;
-+ struct dentry *regs_info;
-+};
-+
-+#endif
Modified: branches/src/target/kernel/2.6.23.x/patches/series
===================================================================
--- branches/src/target/kernel/2.6.23.x/patches/series 2007-09-26 16:12:59 UTC (rev 3046)
+++ branches/src/target/kernel/2.6.23.x/patches/series 2007-09-26 16:16:04 UTC (rev 3047)
@@ -1,4 +1,3 @@
-#alsa-2.6.23-rc1-commit.dif (merged upstream)
asoc-platform-hw_init-pcm_emulation-fix.patch
asoc-kconfig-fix.patch
missing_defs.patch
@@ -12,7 +11,6 @@
#s3c2410_udc_from_upstream.p (merged upstream)
s3c2410_touchscreen.patch
s3c2410-bbt.patch
-#udc-nomodule-misccr.patch (already dropped for 2.6.22.x)
s3c_mci.patch
s3cmci_dbg.patch
s3cmci-dma-free.patch
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