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authorLinus Torvalds <torvalds@woody.linux-foundation.org>2007-07-20 08:52:06 -0700
committerLinus Torvalds <torvalds@woody.linux-foundation.org>2007-07-20 08:52:06 -0700
commit7f46e6ca0183568a688e6bfe40e3ab9adb305d03 (patch)
tree32934472c0396953c910e1dd14c195c782ae9986
parentc8e16aa291c7d1ff14d3784e52bd376bcae17d54 (diff)
parent53555eb758388fff3982875129ca96080caed147 (diff)
Merge branch 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa
* 'linus' of master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (102 commits) [ALSA] version 1.0.14 [ALSA] remove duplicate Logitech Quickcam USB ID in usbquirks.h [ALSA] hda-codec - Fix input with STAC92xx [ALSA] hda-intel: support for iMac 24'' released on 09/2006 [ALSA] hda-codec - Add quirk for Asus P5LD2 [ALSA] snd-ca0106: Add support for X-Fi Extreme Audio. [ALSA] snd-emu10k1:Enable E-Mu 1616m notebook firmware loading. [ALSA] snd-emu10k1: Initial support for E-Mu 1616 and 1616m. [ALSA] cs46xx - Fix PM resume [ALSA] hda: Enable SPDIF in/out on some stac9205 boards [ALSA] timer: check for incorrect device state in non-debug compiles, too [ALSA] snd-aoa-codec-onyx: fix typo [ALSA] hda-codec - Add quirks for HP dx2200/dx2250 [ALSA] hda-codec - Rename HP model-specific quirks [ALSA] hda-codec - Add quirk for HP Samba [ALSA] hda-codec - Add LG LW20 line-in capture source [ALSA] usb-audio - Fix AC3 with M-Audio Audiophile USB [ALSA] hda: stac9202 mixer fix [ALSA] Make s3c24xx_i2s_set_clkdiv() change the correct bits [ALSA] hda-codec - Add LG LW20 si3054 modem id ...
-rw-r--r--CREDITS10
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt75
-rw-r--r--Documentation/sound/alsa/Audiophile-Usb.txt242
-rw-r--r--Documentation/sound/alsa/OSS-Emulation.txt15
-rw-r--r--include/linux/i2c-id.h7
-rw-r--r--include/sound/ak4xxx-adda.h1
-rw-r--r--include/sound/cs46xx.h4
-rw-r--r--include/sound/cs46xx_dsp_spos.h2
-rw-r--r--include/sound/emu10k1.h16
-rw-r--r--include/sound/sb.h1
-rw-r--r--include/sound/version.h2
-rw-r--r--include/sound/wavefront_fx.h9
-rw-r--r--sound/Kconfig2
-rw-r--r--sound/Makefile2
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-onyx.c4
-rw-r--r--sound/core/pcm_native.c2
-rw-r--r--sound/core/seq/seq_instr.c6
-rw-r--r--sound/core/timer.c27
-rw-r--r--sound/drivers/dummy.c2
-rw-r--r--sound/drivers/mpu401/mpu401.c2
-rw-r--r--sound/drivers/portman2x4.c2
-rw-r--r--sound/drivers/serial-u16550.c2
-rw-r--r--sound/drivers/virmidi.c2
-rw-r--r--sound/i2c/other/ak4xxx-adda.c24
-rw-r--r--sound/isa/Kconfig32
-rw-r--r--sound/isa/ad1848/ad1848_lib.c4
-rw-r--r--sound/isa/opl3sa2.c2
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c3
-rw-r--r--sound/isa/sb/Makefile15
-rw-r--r--sound/isa/sb/sb16_main.c10
-rw-r--r--sound/isa/sb/sb_common.c5
-rw-r--r--sound/isa/sb/sb_mixer.c3
-rw-r--r--sound/isa/sscape.c4
-rw-r--r--sound/isa/wavefront/wavefront_synth.c2
-rw-r--r--sound/pci/Kconfig11
-rw-r--r--sound/pci/Makefile2
-rw-r--r--sound/pci/ali5451/ali5451.c7
-rw-r--r--sound/pci/als300.c7
-rw-r--r--sound/pci/ca0106/ca0106_main.c19
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c77
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.h3
-rw-r--r--sound/pci/cs46xx/dsp_spos.c170
-rw-r--r--sound/pci/cs5530.c306
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c125
-rw-r--r--sound/pci/emu10k1/emufx.c78
-rw-r--r--sound/pci/emu10k1/emumixer.c16
-rw-r--r--sound/pci/emu10k1/emupcm.c39
-rw-r--r--sound/pci/ens1370.c4
-rw-r--r--sound/pci/hda/hda_intel.c53
-rw-r--r--sound/pci/hda/hda_proc.c6
-rw-r--r--sound/pci/hda/patch_analog.c630
-rw-r--r--sound/pci/hda/patch_atihdmi.c1
-rw-r--r--sound/pci/hda/patch_conexant.c2
-rw-r--r--sound/pci/hda/patch_realtek.c919
-rw-r--r--sound/pci/hda/patch_si3054.c4
-rw-r--r--sound/pci/hda/patch_sigmatel.c266
-rw-r--r--sound/pci/ice1712/revo.c7
-rw-r--r--sound/pci/nm256/nm256.c3
-rw-r--r--sound/pci/rme9652/rme9652.c2
-rw-r--r--sound/pci/via82xx.c4
-rw-r--r--sound/pci/via82xx_modem.c4
-rw-r--r--sound/ppc/Kconfig20
-rw-r--r--sound/ppc/Makefile3
-rw-r--r--sound/ppc/snd_ps3.c1125
-rw-r--r--sound/ppc/snd_ps3.h135
-rw-r--r--sound/ppc/snd_ps3_reg.h891
-rw-r--r--sound/sh/Kconfig14
-rw-r--r--sound/sh/Makefile8
-rw-r--r--sound/sh/aica.c665
-rw-r--r--sound/sh/aica.h81
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile2
-rw-r--r--sound/soc/s3c24xx/Kconfig27
-rw-r--r--sound/soc/s3c24xx/Makefile9
-rw-r--r--sound/soc/s3c24xx/lm4857.h32
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c670
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c401
-rw-r--r--sound/soc/s3c24xx/s3c24xx-ac97.h25
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c4
-rw-r--r--sound/soc/s3c24xx/smdk2443_wm9710.c85
-rw-r--r--sound/soc/sh/Kconfig38
-rw-r--r--sound/soc/sh/Makefile14
-rw-r--r--sound/soc/sh/dma-sh7760.c354
-rw-r--r--sound/soc/sh/hac.c322
-rw-r--r--sound/soc/sh/sh7760-ac97.c92
-rw-r--r--sound/soc/sh/ssi.c400
-rw-r--r--sound/usb/usbaudio.c22
-rw-r--r--sound/usb/usbquirks.h72
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c7
89 files changed, 8426 insertions, 399 deletions
diff --git a/CREDITS b/CREDITS
index 79fd13dbb8e..10c214dc95e 100644
--- a/CREDITS
+++ b/CREDITS
@@ -2212,13 +2212,13 @@ S: 2300 Copenhagen S
S: Denmark
N: Claudio S. Matsuoka
-E: claudio@conectiva.com
-E: claudio@helllabs.org
+E: cmatsuoka@gmail.com
+E: claudio@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
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 355ff0a2bb7..241e26c4ff9 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
above explicitly.
The power-management is supported.
-
+
+ Module snd-cs5530
+ _________________
+
+ Module for Cyrix/NatSemi Geode 5530 chip.
+
Module snd-cs5535audio
----------------------
@@ -759,6 +764,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)
+ probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
single_cmd - Use single immediate commands to communicate with
codecs (for debugging only)
enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
@@ -803,6 +809,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
hp-3013 HP machines (3013-variant)
fujitsu Fujitsu S7020
acer Acer TravelMate
+ will Will laptops (PB V7900)
+ replacer Replacer 672V
basic fixed pin assignment (old default model)
auto auto-config reading BIOS (default)
@@ -811,16 +819,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
hp-bpc HP xw4400/6400/8400/9400 laptops
hp-bpc-d7000 HP BPC D7000
benq Benq ED8
+ benq-t31 Benq T31
hippo Hippo (ATI) with jack detection, Sony UX-90s
hippo_1 Hippo (Benq) with jack detection
+ sony-assamd Sony ASSAMD
basic fixed pin assignment w/o SPDIF
auto auto-config reading BIOS (default)
+ ALC268
+ 3stack 3-stack model
+ auto auto-config reading BIOS (default)
+
+ ALC662
+ 3stack-dig 3-stack (2-channel) with SPDIF
+ 3stack-6ch 3-stack (6-channel)
+ 3stack-6ch-dig 3-stack (6-channel) with SPDIF
+ 6stack-dig 6-stack with SPDIF
+ lenovo-101e Lenovo laptop
+ auto auto-config reading BIOS (default)
+
ALC882/885
3stack-dig 3-jack with SPDIF I/O
6stack-dig 6-jack digital with SPDIF I/O
arima Arima W820Di1
macpro MacPro support
+ imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
auto auto-config reading BIOS (default)
@@ -832,9 +855,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
6stack-dig-demo 6-jack digital for Intel demo board
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
medion Medion Laptops
+ medion-md2 Medion MD2
targa-dig Targa/MSI
targa-2ch-dig Targs/MSI with 2-channel
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
+ lenovo-101e Lenovo 101E
+ lenovo-nb0763 Lenovo NB0763
+ lenovo-ms7195-dig Lenovo MS7195
+ 6stack-hp HP machines with 6stack (Nettle boards)
+ 3stack-hp HP machines with 3stack (Lucknow, Samba boards)
auto auto-config reading BIOS (default)
ALC861/660
@@ -853,7 +882,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
3stack-dig 3-jack with SPDIF OUT
6stack-dig 6-jack with SPDIF OUT
3stack-660 3-jack (for ALC660VD)
+ 3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
lenovo Lenovo 3000 C200
+ dallas Dallas laptops
auto auto-config reading BIOS (default)
CMI9880
@@ -864,12 +895,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
allout 5-jack in back, 2-jack in front, SPDIF out
auto auto-config reading BIOS (default)
+ AD1882
+ 3stack 3-stack mode (default)
+ 6stack 6-stack mode
+
+ AD1884
+ N/A
+
AD1981
basic 3-jack (default)
hp HP nx6320
thinkpad Lenovo Thinkpad T60/X60/Z60
toshiba Toshiba U205
+ AD1983
+ N/A
+
+ AD1984
+ basic default configuration
+ thinkpad Lenovo Thinkpad T61/X61
+
AD1986A
6stack 6-jack, separate surrounds (default)
3stack 3-stack, shared surrounds
@@ -907,11 +952,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
ref Reference board
3stack D945 3stack
5stack D945 5stack + SPDIF
- macmini Intel Mac Mini
- macbook Intel Mac Book
- macbook-pro-v1 Intel Mac Book Pro 1st generation
- macbook-pro Intel Mac Book Pro 2nd generation
- imac-intel Intel iMac
+ dell Dell XPS M1210
+ intel-mac-v1 Intel Mac Type 1
+ intel-mac-v2 Intel Mac Type 2
+ intel-mac-v3 Intel Mac Type 3
+ intel-mac-v4 Intel Mac Type 4
+ intel-mac-v5 Intel Mac Type 5
+ macmini Intel Mac Mini (equivalent with type 3)
+ macbook Intel Mac Book (eq. type 5)
+ macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+ macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3)
+ imac-intel Intel iMac (eq. type 2)
+ imac-intel-20 Intel iMac (newer version) (eq. type 3)
STAC9202/9250/9251
ref Reference board, base config
@@ -956,6 +1008,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
from the irq. Remember this is a last resort, and should be
avoided as much as possible...
+ MORE NOTES ON "azx_get_response timeout" PROBLEMS:
+ On some hardwares, you may need to add a proper probe_mask option
+ to avoid the "azx_get_response timeout" problem above, instead.
+ This occurs when the access to non-existing or non-working codec slot
+ (likely a modem one) causes a stall of the communication via HD-audio
+ bus. You can see which codec slots are probed by enabling
+ CONFIG_SND_DEBUG_DETECT, or simply from the file name of the codec
+ proc files. Then limit the slots to probe by probe_mask option.
+ For example, probe_mask=1 means to probe only the first slot, and
+ probe_mask=4 means only the third slot.
+
The power-management is supported.
Module snd-hdsp
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index e40cce83327..2ad5e6306c4 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -1,4 +1,4 @@
- Guide to using M-Audio Audiophile USB with ALSA and Jack v1.3
+ Guide to using M-Audio Audiophile USB with ALSA and Jack v1.5
========================================================
Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -6,8 +6,19 @@
This document is a guide to using the M-Audio Audiophile USB (tm) device with
ALSA and JACK.
+History
+=======
+* v1.4 - Thibault Le Meur (2007-07-11)
+ - Added Low Endianness nature of 16bits-modes
+ found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+ - Modifying document structure
+* v1.5 - Thibault Le Meur (2007-07-12)
+ - Added AC3/DTS passthru info
+
+
1 - Audiophile USB Specs and correct usage
==========================================
+
This part is a reminder of important facts about the functions and limitations
of the device.
@@ -25,18 +36,18 @@ The device has 4 audio interfaces, and 2 MIDI ports:
The internal DAC/ADC has the following characteristics:
* sample depth of 16 or 24 bits
* sample rate from 8kHz to 96kHz
-* Two ports can't use different sample depths at the same time. Moreover, the
-Audiophile USB documentation gives the following Warning: "Please exit any
-audio application running before switching between bit depths"
+* Two interfaces can't use different sample depths at the same time.
+Moreover, the Audiophile USB documentation gives the following Warning:
+"Please exit any audio application running before switching between bit depths"
Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be
activated at the same time depending on the audio mode selected:
- * 16-bit/48kHz ==> 4 channels in/4 channels out
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
- Ai+Ao+Di+Do
- * 24-bit/48kHz ==> 4 channels in/2 channels out,
- or 2 channels in/4 channels out
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out,
+ or 2 channels in + 4 channels out
- Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
- * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
- Ai or Ao or Di or Do
Important facts about the Digital interface:
@@ -52,44 +63,56 @@ source is connected
synchronization error (for instance sound played at an odd sample rate)
-2 - Audiophile USB support in ALSA
-==================================
+2 - Audiophile USB MIDI support in ALSA
+=======================================
-2.1 - MIDI ports
-----------------
-The Audiophile USB MIDI ports will be automatically supported once the
+The Audiophile USB MIDI ports will be automatically supported once the
following modules have been loaded:
* snd-usb-audio
* snd-seq-midi
No additional setting is required.
-2.2 - Audio ports
------------------
+
+3 - Audiophile USB Audio support in ALSA
+========================================
Audio functions of the Audiophile USB device are handled by the snd-usb-audio
module. This module can work in a default mode (without any device-specific
parameter), or in an "advanced" mode with the device-specific parameter called
"device_setup".
-2.2.1 - Default Alsa driver mode
-
-The default behavior of the snd-usb-audio driver is to parse the device
-capabilities at startup and enable all functions inside the device (including
-all ports at any supported sample rates and sample depths). This approach
-has the advantage to let the driver easily switch from sample rates/depths
-automatically according to the need of the application claiming the device.
-
-In this case the Audiophile ports are mapped to alsa pcm devices in the
-following way (I suppose the device's index is 1):
+3.1 - Default Alsa driver mode
+------------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device
+capabilities at startup and activate the required mode when required
+by the applications: for instance if the user is recording in a
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample
+rates/depths automatically according to the user's needs. However, those who
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now. In this case the
+Audiophile interfaces are mapped to alsa pcm devices in the following
+way (I suppose the device's index is 1):
* hw:1,0 is Ao in playback and Di in capture
* hw:1,1 is Do in playback and Ai in capture
* hw:1,2 is Do in AC3/DTS passthrough mode
-You must note as well that the device uses Big Endian byte encoding so that
-supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
-24-bits depth mode. One exception is the hw:1,2 port which is Little Endian
-compliant and thus uses S16_LE.
+In this mode, the device uses Big Endian byte-encoding so that
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface
+is reported to be big endian in this default driver mode.
Examples:
* playing a S24_3BE encoded raw file to the Ao port
@@ -98,22 +121,26 @@ Examples:
% arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw
* playing a S16_BE encoded raw file to the Do port
% aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+ * playing an ac3 sample file to the Do port
+ % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
-If you're happy with the default Alsa driver setup and don't experience any
+If you're happy with the default Alsa driver mode and don't experience any
issue with this mode, then you can skip the following chapter.
-2.2.2 - Advanced module setup
+3.2 - Advanced module setup
+---------------------------
Due to the hardware constraints described above, the device initialization made
by the Alsa driver in default mode may result in a corrupted state of the
device. For instance, a particularly annoying issue is that the sound captured
-from the Ai port sounds distorted (as if boosted with an excessive high volume
-gain).
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
For people having this problem, the snd-usb-audio module has a new module
-parameter called "device_setup".
+parameter called "device_setup" (this parameter was introduced in kernel
+release 2.6.17)
-2.2.2.1 - Initializing the working mode of the Audiophile USB
+3.2.1 - Initializing the working mode of the Audiophile USB
As far as the Audiophile USB device is concerned, this value let the user
specify:
@@ -121,33 +148,57 @@ specify:
* the sample rate
* whether the Di port is used or not
-Here is a list of supported device_setup values for this device:
- * device_setup=0x00 (or omitted)
- - Alsa driver default mode
- - maintains backward compatibility with setups that do not use this
- parameter by not introducing any change
- - results sometimes in corrupted sound as described earlier
+When initialized with "device_setup=0x00", the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+3.2.1.1 - 16-bit modes
+
+The two supported modes are:
+
* device_setup=0x01
- 16bits 48kHz mode with Di disabled
- Ai,Ao,Do can be used at the same time
- hw:1,0 is not available in capture mode
- hw:1,2 is not available
+
* device_setup=0x11
- 16bits 48kHz mode with Di enabled
- Ai,Ao,Di,Do can be used at the same time
- hw:1,0 is available in capture mode
- hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+ % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+3.2.1.2 - 24-bit modes
+
+The three supported modes are:
+
* device_setup=0x09
- 24bits 48kHz mode with Di disabled
- Ai,Ao,Do can be used at the same time
- hw:1,0 is not available in capture mode
- hw:1,2 is not available
+
* device_setup=0x19
- 24bits 48kHz mode with Di enabled
- 3 ports from {Ai,Ao,Di,Do} can be used at the same time
- hw:1,0 is available in capture mode and an active digital source must be
connected to Di
- hw:1,2 is not available
+
* device_setup=0x0D or 0x10
- 24bits 96kHz mode
- Di is enabled by default for this mode but does not need to be connected
@@ -155,34 +206,64 @@ Here is a list of supported device_setup values for this device:
- Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
- hw:1,0 is available in captured mode
- hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver
+mode" above for an aplay command example)
+
+3.2.1.3 - AC3 w/ DTS passthru mode
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
* device_setup=0x03
- 16bits 48kHz mode with only the Do port enabled
- - AC3 with DTS passthru (not tested)
+ - AC3 with DTS passthru
- Caution with this setup the Do port is mapped to the pcm device hw:1,0
-2.2.2.2 - Setting and switching configurations with the device_setup parameter
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+ % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+3.2.2 - How to use the device_setup parameter
+----------------------------------------------
The parameter can be given:
+
* By manually probing the device (as root):
# modprobe -r snd-usb-audio
# modprobe snd-usb-audio index=1 device_setup=0x09
+
* Or while configuring the modules options in your modules configuration file
- For Fedora distributions, edit the /etc/modprobe.conf file:
alias snd-card-1 snd-usb-audio
options snd-usb-audio index=1 device_setup=0x09
-IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
--------------------------------------------
- * You may need to _first_ initialize the module with the correct device_setup
- parameter and _only_after_ turn on the Audiophile USB device
- * This is especially true when switching the sample depth:
+CAUTION when initializaing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+ the module BEFORE the device is turned on. So, if you use the "manual probing"
+ method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead in a misconfiguration of the device. In this case
+ turn off the device, unproble the snd-usb-audio module, then probe it again with
+ correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+ to another mode (possibly with another sample-depth), please use also the following
+ procedure:
- first turn off the device
- de-register the snd-usb-audio module (modprobe -r)
- change the device_setup parameter by changing the device_setup
option in /etc/modprobe.conf
- turn on the device
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+ be enough to ensure the 'stability' of the device initialization.
-2.2.2.3 - Audiophile USB's device_setup structure
+3.2.3 - Technical details for hackers
+-------------------------------------
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+3.2.3.1 - Audiophile USB's device_setup structure
If you want to understand the device_setup magic numbers for the Audiophile
USB, you need some very basic understanding of binary computation. However,
@@ -228,12 +309,12 @@ Caution:
- choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
only be able to use one at the same time
-2.2.3 - USB implementation details for this device
+3.2.3.2 - USB implementation details for this device
You may safely skip this section if you're not interested in driver
-development.
+hacking.
-This section describes some internal aspects of the device and summarize the
+This section describes some internal aspects of the device and summarizes the
data I got by usb-snooping the windows and Linux drivers.
The M-Audio Audiophile USB has 7 USB Interfaces:
@@ -293,43 +374,45 @@ parse_audio_endpoints function uses a quirk called
"audiophile_skip_setting_quirk" in order to prevent AltSettings not
corresponding to device_setup from being registered in the driver.
-3 - Audiophile USB and Jack support
+4 - Audiophile USB and Jack support
===================================
This section deals with support of the Audiophile USB device in Jack.
-The main issue regarding this support is that the device is Big Endian
-compliant.
-3.1 - Using the plug alsa plugin
---------------------------------
+There are 2 main potential issues when using Jackd with the device:
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+4.1 - Direct support in Jackd
+-----------------------------
-Jack doesn't directly support big endian devices. Thus, one way to have support
-for this device with Alsa is to use the Alsa "plug" converter.
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember
+extacly when this support was released into jackd, let's just say that
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting
+16bits Big-Endian devices, but since you've read carefully the above
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+ % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+4.2 - Using Alsa plughw
+-----------------------
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa "plug" converter.
For instance here is one way to run Jack with 2 playback channels on Ao and 2
capture channels from Ai:
% jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
-
However you may see the following warning message:
"You appear to be using the ALSA software "plug" layer, probably a result of
using the "default" ALSA device. This is less efficient than it could be.
Consider using a hardware device instead rather than using the plug layer."
-3.2 - Patching alsa to use direct pcm device
---------------------------------------------
-A patch for Jack by Andreas Steinmetz adds support for Big Endian devices.
-However it has not been included in the CVS tree.
-
-You can find it at the following URL:
-http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&
-atid=425939
-
-After having applied the patch you can run jackd with the following command
-line:
- % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
-
-3.2 - Getting 2 input and/or output interfaces in Jack
+4.3 - Getting 2 input and/or output interfaces in Jack
------------------------------------------------------
As you can see, starting the Jack server this way will only enable 1 stereo
@@ -339,6 +422,7 @@ This is due to the following restrictions:
* Jack can only open one capture device and one playback device at a time
* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
(and optionally hw:1,2)
+
If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
combine the Alsa devices into one logical "complex" device.
@@ -348,13 +432,11 @@ It is related to another device (ice1712) but can be adapted to suit
the Audiophile USB.
Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
-* patching Jack with the previously mentioned "Big Endian" patch
-* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
-* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
file
* start jackd with this device
-I had no success in testing this for now, but this may be due to my OS
-configuration. If you have any success with this kind of setup, please
-drop me an email.
+I had no success in testing this for now, if you have any success with this kind
+of setup, please drop me an email.
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
index ec2a02541d5..bfa0c9aacb4 100644
--- a/Documentation/sound/alsa/OSS-Emulation.txt
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -278,6 +278,21 @@ current mixer configuration by reading and writing the whole file
image.
+Duplex Streams
+==============
+
+Note that when attempting to use a single device file for playback and
+capture, the OSS API provides no way to set the format, sample rate or
+number of channels different in each direction. Thus
+ io_handle = open("device", O_RDWR)
+will only function correctly if the values are the same in each direction.
+
+To use different values in the two directions, use both
+ input_handle = open("device", O_RDONLY)
+ output_handle = open("device", O_WRONLY)
+and set the values for the corresponding handle.
+
+
Unsupported Features
====================
diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h
index aa83d416309..b6901486571 100644
--- a/include/linux/i2c-id.h
+++ b/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 */
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
index aa49dda4f41..fd0a6c46f49 100644
--- a/include/sound/ak4xxx-adda.h
+++ b/include/sound/ak4xxx-adda.h
@@ -43,6 +43,7 @@ struct snd_ak4xxx_ops {
struct snd_akm4xxx_dac_channel {
char *name; /* mixer volume name */
unsigned int num_channels;
+ char *switch_name; /* mixer switch*/
};
/* ADC labels and channels */
diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h
index 685928e6f65..353910ce975 100644
--- a/include/sound/cs46xx.h
+++ b/include/sound/cs46xx.h
@@ -1723,6 +1723,10 @@ struct snd_cs46xx {
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,
diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h
index da934def31e..d9da9e59cf3 100644
--- a/include/sound/cs46xx_dsp_spos.h
+++ b/include/sound/cs46xx_dsp_spos.h
@@ -107,6 +107,7 @@ struct dsp_scb_descriptor {
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 @@ struct dsp_task_descriptor {
int size;
u32 address;
int index;
+ u32 *data;
};
struct dsp_pcm_channel_descriptor {
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 23e45a4cf0e..529d0a56436 100644
--- a/include/sound/emu10k1.h
+++ b/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 */
diff --git a/include/sound/sb.h b/include/sound/sb.h
index 2dd5c8e5b4f..3ad854b397d 100644
--- a/include/sound/sb.h
+++ b/include/sound/sb.h
@@ -38,6 +38,7 @@ enum sb_hw_type {
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
diff --git a/include/sound/version.h b/include/sound/version.h
index 8e5b2f0f594..6bbcfefd2c3 100644
--- a/include/sound/version.h
+++ b/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)"
diff --git a/include/sound/wavefront_fx.h b/include/sound/wavefront_fx.h
deleted file mode 100644
index cec92b14179..00000000000
--- a/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
diff --git a/sound/Kconfig b/sound/Kconfig
index 9ea47382341..e48b9b37d22 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -65,6 +65,8 @@ source "sound/arm/Kconfig"
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"
diff --git a/sound/Makefile b/sound/Makefile
index b7c7fb7c24c..3ead922bd9c 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o
obj-$(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
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c
index ded51671794..028852374f2 100644
--- a/sound/aoa/codecs/snd-aoa-codec-onyx.c
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
@@ -661,7 +661,7 @@ static struct transfer_info onyx_transfers[] = {
.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 @@ static int onyx_prepare(struct codec_info_item *cii,
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;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a96733a5beb..59b29cd482a 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1487,7 +1487,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream)
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 */
diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c
index f30d171b6d9..5efe6523a58 100644
--- a/sound/core/seq/seq_instr.c
+++ b/sound/core/seq/seq_instr.c
@@ -109,7 +109,7 @@ void snd_seq_instr_list_free(struct snd_seq_kinstr_list **list_ptr)
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 @@ int snd_seq_instr_list_free_cond(struct snd_seq_kinstr_list *list,
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 @@ static int instr_free(struct snd_seq_kinstr_ops *ops,
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);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 67520b3c004..f2bbacedd56 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1549,9 +1549,11 @@ static int snd_timer_user_info(struct file *file,
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 @@ static int snd_timer_user_params(struct file *file,
int err;
tu = file->private_data;
- snd_assert(tu->timeri != NULL, return -ENXIO);
+ if (!tu->timeri)
+ return -EBADFD;
t = tu->timeri->timer;
- snd_assert(t != NULL, return -ENXIO);
+ if (!t)
+ return -EBADFD;
if (copy_from_user(&params, _params, sizeof(params)))
return -EFAULT;
if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) {
@@ -1675,7 +1679,8 @@ static int snd_timer_user_status(struct file *file,
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 @@ static int snd_timer_user_start(struct file *file)
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 @@ static int snd_timer_user_stop(struct file *file)
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 @@ static int snd_timer_user_continue(struct file *file)
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 @@ static int snd_timer_user_pause(struct file *file)
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;
}
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index a0f28f51fc7..4360ae9de19 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -659,7 +659,7 @@ static struct platform_driver snd_dummy_driver = {
},
};
-static void __init_or_module snd_dummy_unregister_all(void)
+static void snd_dummy_unregister_all(void)
{
int i;
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 1d563e515c1..67c6e974541 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -228,7 +228,7 @@ static struct pnp_driver snd_mpu401_pnp_driver = {
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;
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 497cafb57d9..0eb9b5cebfc 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -833,7 +833,7 @@ static struct platform_driver snd_portman_driver = {
/*********************************************************************
* module init stuff
*********************************************************************/
-static void __init_or_module snd_portman_unregister_all(void)
+static void snd_portman_unregister_all(void)
{
int i;
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index 838a4277929..d3e6a20edd3 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -998,7 +998,7 @@ static struct platform_driver snd_serial_driver = {
},
};
-static void __init_or_module snd_serial_unregister_all(void)
+static void snd_serial_unregister_all(void)
{
int i;
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
index 46f3d348606..915c86773c2 100644
--- a/sound/drivers/virmidi.c
+++ b/sound/drivers/virmidi.c
@@ -145,7 +145,7 @@ static struct platform_driver snd_virmidi_driver = {
},
};
-static void __init_or_module snd_virmidi_unregister_all(void)
+static void snd_virmidi_unregister_all(void)
{
int i;
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 8805110017a..fd335159f84 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -481,8 +481,8 @@ static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol,
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 @@ static int build_dac_controls(struct snd_akm4xxx *ak)
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";
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index cf3803cd579..ea5084abe60 100644
--- a/sound/isa/Kconfig
+++ b/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 @@ config SND_CS4231_LIB
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 @@ config SND_ALS100
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 @@ config SND_CMI8330
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 @@ config SND_DT019X
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 @@ config SND_ES968
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 @@ config SND_SB8
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 @@ config SND_SB16
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 @@ config SND_SBAWE
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).
diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c
index 8094282c2ae..1bc2e3fd572 100644
--- a/sound/isa/ad1848/ad1848_lib.c
+++ b/sound/isa/ad1848/ad1848_lib.c
@@ -245,7 +245,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip)
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 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip)
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);
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 4f6800b43b0..e70db32991d 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -164,6 +164,8 @@ static struct pnp_card_device_id snd_opl3sa2_pnpids[] = {
{ .id = "YMH0801", .devs = { { "YMH0021" } } },
/* NeoMagic MagicWave 3DX */
{ .id = "NMX2200", .devs = { { "YMH2210" } } },
+ /* NeoMagic MagicWave 3D */
+ { .id = "NMX2200", .devs = { { "NMX2210" } } },
/* --- */
{ .id = "" } /* end */
};
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 60c120ffb9d..049d479ce2b 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -1927,10 +1927,12 @@ static struct snd_card *snd_opti9xx_card_new(void)
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 @@ static int __init alsa_card_opti9xx_init(void)
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);
}
diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile
index fd9d9c5726f..556e6692802 100644
--- a/sound/isa/sb/Makefile
+++ b/sound/isa/sb/Makefile
@@ -22,14 +22,13 @@ snd-es968-objs := es968.o
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
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
index 383911b9e74..5d4d3aafe2d 100644
--- a/sound/isa/sb/sb16_main.c
+++ b/sound/isa/sb/sb16_main.c
@@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct snd_pcm_substream *substream)
__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 @@ static int snd_sb16_capture_open(struct snd_pcm_substream *substream)
__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;
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index 3094f385216..efa9d5c2558 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb * chip)
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 @@ static int snd_sbdsp_probe(struct snd_sb * chip)
case SB_HW_DT019X:
str = "(DT019X/ALS007)";
break;
+ case SB_HW_CS5530:
+ str = "16 (CS5530)";
+ break;
default:
return -ENODEV;
}
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 490b1ca5cf5..3d4befcff28 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip)
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 @@ void snd_sbmixer_suspend(struct snd_sb *chip)
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 @@ void snd_sbmixer_resume(struct snd_sb *chip)
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:
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 9ea417bcf3e..cbad2a51cba 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -382,7 +382,7 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout)
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 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout)
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));
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 78020d832e0..bacc51c8658 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1780,7 +1780,7 @@ wavefront_should_cause_interrupt (snd_wavefront_t *dev,
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;
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 61e35ecc57b..c6b44102aa5 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -33,6 +33,7 @@ config SND_ALS4000
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 @@ config SND_CS46XX_NEW_DSP
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
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index e06736da9ef..cd76e0293d0 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -12,6 +12,7 @@ snd-azt3328-objs := azt3328.o
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-$(CONFIG_SND_AZT3328) += snd-azt3328.o
obj-$(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
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 41543a4933e..05b4c869694 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -239,7 +239,7 @@ struct snd_ali_image {
struct snd_ali {
- unsigned long irq;
+ int irq;
unsigned long port;
unsigned char revision;
@@ -731,8 +731,7 @@ static void snd_ali_detect_spdif_rate(struct snd_ali *codec)
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 @@ static int __devinit snd_ali_probe(struct pci_dev *pci,
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");
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8afcb98ca7b..48cc39b771d 100644
--- a/sound/pci/als300.c
+++ b/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 @@ static int __devinit snd_als300_create(struct snd_card *card,
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;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 9fd7b8a5b75..fcab8fb97e3 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -168,6 +168,25 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
#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]",
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index bef1f6d1859..71d7aab9d86 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -2897,6 +2897,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip)
}
#endif
+#ifdef CONFIG_PM
+ kfree(chip->saved_regs);
+#endif
+
pci_disable_device(chip->pci);
kfree(chip);
return 0;
@@ -3140,6 +3144,23 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip)
/*
* 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 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip)
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 @@ static struct cs_card_type __devinitdata cards[] = {
* 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 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
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 @@ int snd_cs46xx_resume(struct pci_dev *pci)
{
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 @@ int snd_cs46xx_resume(struct pci_dev *pci)
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 @@ int snd_cs46xx_resume(struct pci_dev *pci)
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 @@ int __devinit snd_cs46xx_create(struct snd_card *card,
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);
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index f75750c2bd2..20dcd72f06c 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -86,6 +86,9 @@ static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned
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
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index 336e77e2600..590b35d91df 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -306,13 +306,59 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip)
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 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m
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 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size)
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 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size)
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 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32
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 @@ cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_da
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 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip)
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 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip,
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 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right)
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
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
new file mode 100644
index 00000000000..240a0a46220
--- /dev/null
+++ b/sound/pci/cs5530.c
@@ -0,0 +1,306 @@
+/*
+ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
+ *
+ * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
+ * (C) Copyright 2003 Red Hat Inc <alan@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)
+
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 4a9b59ad8ab..404ae1be0a4 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/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 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
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 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
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 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
/* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */
snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
snd_printdd("reg1=0x%x\n",reg);
- if (reg == 0x55) {
+ if ((reg & 0x3f) == 0x15) {
/* FPGA netlist already present so clear it */
/* Return to programming mode */
@@ -735,19 +774,32 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
}
snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
snd_printdd("reg2=0x%x\n",reg);
- if (reg == 0x55) {
+ if ((reg & 0x3f) == 0x15) {
/* FPGA failed to return to programming mode */
+ snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n");
return -ENODEV;
}
snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg);
- if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) {
- snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME);
- return err;
+ if (emu->card_capabilities->emu1010 == 1) {
+ if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) {
+ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME);
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 2) {
+ if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) {
+ snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME);
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 3) {
+ if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) {
+ snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME);
+ return err;
+ }
}
/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
- if (reg != 0x55) {
+ if ((reg & 0x3f) != 0x15) {
/* FPGA failed to be programmed */
snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg);
return -ENODEV;
@@ -850,6 +902,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
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 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
/* Return to Audio Dock programming mode */
snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
- if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
- return err;
+ if (emu->card_capabilities->emu1010 == 1) {
+ if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 2) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 3) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
}
+
snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
/* ID, should read & 0x7f = 0x55 when FPGA programmed. */
snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
- if (reg != 0x55) {
+ if ((reg & 0x3f) != 0x15) {
/* FPGA failed to be programmed */
snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
return 0;
@@ -1227,9 +1311,15 @@ static struct snd_emu_chip_details emu_chip_details[] = {
.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",
@@ -1663,12 +1753,13 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
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);
@@ -1814,10 +1905,10 @@ void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu)
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
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index c02012cccd8..7206c0fa06f 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1123,6 +1123,11 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl
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 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu)
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 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
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 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
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 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
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++) {
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 4db6e1ca166..7b2c1dcc533 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -77,6 +77,10 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol,
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 @@ static char *emu1010_src_texts[] = {
"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 @@ static unsigned int emu1010_src_regs[] = {
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 @@ static unsigned int emu1010_output_dst[] = {
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,
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index ab4f5df5241..eda5cb373de 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1233,24 +1233,26 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
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 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
/* 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 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
#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 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s
/* 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;
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 7c403965153..21cb4268a59 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1607,8 +1607,8 @@ struct es1371_quirk {
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 &&
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2fa281cbef9..92bc8b3fa2a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -341,6 +341,9 @@ struct azx {
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 @@ static void azx_free_cmd_io(struct azx *chip)
}
/* 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 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
}
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 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
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 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
*/
/* 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 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid,
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 @@ static struct pci_device_id azx_ids[] = {
{ 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);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index e313e685f16..ac15066fd30 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -250,6 +250,12 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
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");
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 0e1a879663f..4d7f8d11ad7 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/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@suse.de>
+ * Copyright (c) 2005-2007 Takashi Iwai <tiwai@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 @@ struct ad198x_spec {
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 @@ static int patch_ad1988(struct hda_codec *codec)
/*
+ * 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 },
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index f8eb4c90f80..72d3ab9751a 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -172,6 +172,7 @@ static int patch_atihdmi(struct hda_codec *codec)
*/
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 */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index bef214bcddd..4d8e8af5c81 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -801,7 +801,9 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
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),
{}
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4776de93928..9a47eec5a27 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -94,10 +94,18 @@ enum {
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 @@ enum {
/* ALC861-VD models */
enum {
ALC660VD_3ST,
+ ALC660VD_3ST_DIG,
ALC861VD_3ST,
ALC861VD_3ST_DIG,
ALC861VD_6ST_DIG,
@@ -144,6 +153,7 @@ enum {
ALC882_TARGA,
ALC882_ASUS_A7J,
ALC885_MACPRO,
+ ALC885_IMAC24,
ALC882_AUTO,
ALC882_MODEL_LAST,
};
@@ -163,6 +173,8 @@ enum {
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 @@ static void alc_subsystem_id(struct hda_codec *codec,
}
/*
+ * 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 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
* 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 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = {
{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 @@ static struct alc_config_preset alc880_presets[] = {
.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 @@ static struct hda_verb alc882_macpro_init_verbs[] = {
{ }
};
+/* 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 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
[ALC882_ARIMA] = "arima",
[ALC882_W2JC] = "w2jc",
[ALC885_MACPRO] = "macpro",
+ [ALC885_IMAC24] = "imac24",
[ALC882_AUTO] = "auto",
};
@@ -5284,6 +5404,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
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 @@ static struct alc_config_preset alc882_presets[] = {
.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 @@ static struct alc_config_preset alc882_presets[] = {
/*
+ * 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 @@ static int patch_alc882(struct hda_codec *codec)
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 @@ static int patch_alc882(struct hda_codec *codec)
}
}
+ 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 @@ static int patch_alc882(struct hda_codec *codec)
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 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
{ } /* 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 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
{ } /* 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 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[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 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
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 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
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 @@ static struct alc_config_preset alc883_presets[] = {
.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 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = {
{ } /* 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_EAPD_verbs[] = {
{}
};
+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 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
[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 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
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 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
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 @@ static struct alc_config_preset alc262_presets[] = {
.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,
},
};
@@ -7800,6 +8144,515 @@ static int patch_alc262(struct hda_codec *codec)
}
/*
+ * 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
+};
+
+static hda_nid_t alc268_adc_nids[2] = {
+ /* ADC0-1 */
+ 0x08, 0x07
+};
+
+static hda_nid_t alc268_adc_nids_alt[1] = {
+ /* ADC0 */
+ 0x08
+};
+
+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)
*/
@@ -8767,13 +9620,21 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = {
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 void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
*/
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 const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
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 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
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 @@ static struct alc_config_preset alc861vd_presets[] = {
.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 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
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 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
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 @@ static int patch_alc662(struct hda_codec *codec)
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 @@ static int patch_alc662(struct hda_codec *codec)
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 },
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 43f537ef40b..6d2ecc38905 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -304,8 +304,12 @@ struct hda_codec_preset snd_hda_preset_si3054[] = {
{ .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 },
{}
};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e3964fc4c40..3f25de72966 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -44,6 +44,7 @@ enum {
enum {
STAC_9205_REF,
+ STAC_M43xx,
STAC_9205_MODELS
};
@@ -59,11 +60,19 @@ enum {
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 @@ static hda_nid_t stac9205_pin_nids[12] = {
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 stac9200_mixer[] = {
};
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 @@ static unsigned int d945gtp5_pin_configs[10] = {
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 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
/* 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 ref9205_pin_configs[12] = {
};
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 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
/* 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 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
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 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
* 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 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
}
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 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
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 @@ static int patch_stac9200(struct hda_codec *codec)
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 @@ static int patch_stac925x(struct hda_codec *codec)
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 @@ static int patch_stac922x(struct hda_codec *codec)
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 @@ static int patch_stac922x(struct hda_codec *codec)
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 @@ static int patch_stac927x(struct hda_codec *codec)
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 @@ static int patch_stac927x(struct hda_codec *codec)
}
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 @@ static int patch_stac927x(struct hda_codec *codec)
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 @@ static int patch_stac9205(struct hda_codec *codec)
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 @@ static struct hda_input_mux vaio_mux = {
.num_items = 2,
.items = {
/* { "HP", 0x0 }, */
- { "Line", 0x1 },
- { "Mic", 0x2 },
+ { "Mic Jack", 0x1 },
+ { "Internal Mic", 0x2 },
{ "PCM", 0x3 },
}
};
@@ -2268,7 +2370,7 @@ static struct hda_verb vaio_init[] = {
{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 @@ static struct hda_verb vaio_ar_init[] = {
{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 */
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index 690ceb34064..d18a31e188a 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -186,7 +186,12 @@ static int revo51_i2c_init(struct snd_ice1712 *ice,
#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[] = {
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 03b3a4792f7..c7621bd770a 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1533,7 +1533,8 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci,
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;
}
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index bd7dbd267ed..2de27405a0b 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -406,7 +406,7 @@ static snd_pcm_uframes_t rme9652_hw_pointer(struct snd_rme9652 *rme9652)
} 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) {
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 50c9f92cfd1..6ea09df0c73 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2098,7 +2098,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip)
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 @@ static int snd_via82xx_chip_init(struct via82xx *chip)
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 */
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 8cbf8eba4ae..72425e73aba 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -983,7 +983,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip)
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 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip)
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 */
diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig
index a3fb1496e4d..cacb0b13688 100644
--- a/sound/ppc/Kconfig
+++ b/sound/ppc/Kconfig
@@ -33,3 +33,23 @@ config SND_POWERMAC_AUTO_DRC
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
diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile
index 4d95c652c8c..eacee2d0675 100644
--- a/sound/ppc/Makefile
+++ b/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
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
new file mode 100644
index 00000000000..1aa0b467599
--- /dev/null
+++ b/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);
diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h
new file mode 100644
index 00000000000..4b7e6fbbe50
--- /dev/null
+++ b/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_ */
diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h
new file mode 100644
index 00000000000..03fdee4aaaf
--- /dev/null
+++ b/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
+ */
+
diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig
new file mode 100644
index 00000000000..b7e08ef22a9
--- /dev/null
+++ b/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
+
diff --git a/sound/sh/Makefile b/sound/sh/Makefile
new file mode 100644
index 00000000000..8fdcb6e26f0
--- /dev/null
+++ b/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
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
new file mode 100644
index 00000000000..739786529ca
--- /dev/null
+++ b/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@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@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);
diff --git a/sound/sh/aica.h b/sound/sh/aica.h
new file mode 100644
index 00000000000..8c11e3d10a5
--- /dev/null
+++ b/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@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;
+};
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 10cffc08718..97b25523317 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -27,6 +27,7 @@ config SND_SOC
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"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0ae2e49036f..30414037763 100644
--- a/sound/soc/Makefile
+++ b/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-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 044a3712077..e97c68306a9 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3C24XX chips"
depends on ARCH_S3C2410 && SND_SOC
+ select SND_PCM
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97, I2S or SSP interface. You will also need
@@ -8,3 +9,29 @@ config SND_S3C24XX_SOC
config SND_S3C24XX_SOC_I2S
tristate
+
+config SND_S3C2443_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+config SND_S3C24XX_SOC_NEO1973_WM8753
+ tristate "SoC I2S Audio support for NEO1973 - WM8753"
+ depends on SND_S3C24XX_SOC && MACH_GTA01
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_WM8753
+ help
+ Say Y if you want to add support for SoC audio on smdk2440
+ with the WM8753.
+
+config SND_S3C24XX_SOC_SMDK2443_WM9710
+ tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
+ depends on SND_S3C24XX_SOC && MACH_SMDK2443
+ select SND_S3C2443_SOC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ Say Y if you want to add support for SoC audio on smdk2443
+ with the WM9710.
+
+
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 6f0fffcb30f..13c92f0fa1e 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -1,6 +1,15 @@
# S3c24XX Platform Support
snd-soc-s3c24xx-objs := s3c24xx-pcm.o
snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
+snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
obj-$(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
diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h
new file mode 100644
index 00000000000..0cf5b7011d6
--- /dev/null
+++ b/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@wolfsonmicro.com or linux@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_*/
+
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
new file mode 100644
index 00000000000..d5a8fc2cf8d
--- /dev/null
+++ b/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@wolfsonmicro.com or linux@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@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
new file mode 100644
index 00000000000..75acf7ef552
--- /dev/null
+++ b/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@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * Copyright (C) 2005, Sean Choi <sh428.choi@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");
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
new file mode 100644
index 00000000000..2b835e8260f
--- /dev/null
+++ b/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@wolfsonmicro.com or linux@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_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 8ca314dc889..39f02462e07 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
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;
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
new file mode 100644
index 00000000000..d46cd811ceb
--- /dev/null
+++ b/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@wolfsonmicro.com or linux@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@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
new file mode 100644
index 00000000000..f03220d23e7
--- /dev/null
+++ b/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
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
new file mode 100644
index 00000000000..a8e8ab81cc6
--- /dev/null
+++ b/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
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
new file mode 100644
index 00000000000..cdee374b843
--- /dev/null
+++ b/sound/soc/sh/dma-sh7760.c
@@ -0,0 +1,354 @@
+/*
+ * SH7760 ("camelot") DMABRG audio DMA unit support
+ *
+ * Copyright (C) 2007 Manuel Lauss <mano@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@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
new file mode 100644
index 00000000000..8e3f03908cd
--- /dev/null
+++ b/sound/soc/sh/hac.c
@@ -0,0 +1,322 @@
+/*
+ * Hitachi Audio Controller (AC97) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@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@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
new file mode 100644
index 00000000000..5563f14511f
--- /dev/null
+++ b/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@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
new file mode 100644
index 00000000000..b72bc316cb8
--- /dev/null
+++ b/sound/soc/sh/ssi.c
@@ -0,0 +1,400 @@
+/*
+ * Serial Sound Interface (I2S) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@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@roarinelk.homelinux.net>");
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 8ebc1adb5ed..7bd5852fcc0 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2350,7 +2350,9 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *
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 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *
* 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 snd_usb_cm106_boot_quirk(struct usb_device *dev)
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)
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 374fbf657a2..5a2f518c662 100644
--- a/sound/usb/usbquirks.h
+++ b/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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
}
},
+{
+ 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 @@ YAMAHA_DEVICE(0x7010, "UB99"),
}
},
/* 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 */
{
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 0a352e46862..48e9aa3f18c 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -935,10 +935,9 @@ static struct snd_pcm_ops snd_usX2Y_pcm_ops =
*/
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;
}